Hi List,
I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n,System(echo -e Channel: Channel:
local/23@contest-call\\nContext:
contest-call\\nExtension: 23\\nPriority: 1
Virendra,
you need to change your logic just a bit. in call file a Channel is one
which needs to be dialled fires (See
linkhttp://www.voip-info.org/wiki/view/Asterisk+auto-dial+out).
this will be an extension where your Konference is Hosted for all the other
callers to join. i.e *Channel:
Hey there
You are not moving the call file to spool/outgoing directory. Maybe that's
why you aren't getting anything. I don't feel good about the call file also.
Its not doing what you want it to do.
From: asterisk-users-boun...@lists.digium.com
Hi Sam,
I am doing the same things.
into your suggested script you join into context Konference and then .call
file start IVRs .
the same logic I have pasted in which I make .call file and then join into
the Konference and then .call file start it's work.
But As i know they are on different -2
Hi
1st check that how many manager is connected into the server. 1 or more then
you can say that 2 DTMF is capture by asterisk for same events.
manager show connected
Username IP Address
root 127.0.0.1
it should be one only.
I face the same case then I found that more
On Friday 09 September 2011, bilal ghayyad wrote:
Hi All;
Anyone advise for a free (open source) reporting to be used for asterisk
call center?
Regards
Bilal
Problem is, reporting is such a nebulous thing, about the only thing that will
give you the required level of generality of
sox -h will list the formats supported by your install of sox. If mp3 is not
listed, then your sox does not support mp3. This is not uncommon. Many Linux
distros do not ship support for patent encumbered formats. Either stop using
mp3 (this is what I suggest) or compile and install sox
I don't know why you are running into problems.
Once a call file is executed it creates two legs (according to call file
structure) A leg is Channel: Local/1234@conference and once it Answers the
call file the second leg is bridged which should be
Context-Extension-priority. So what I'm asking is
Dear Tareq;
I am not using mysql, the configuration on the text configuratoin files and the
logs are existed under the directory (/var/log/asterisk).
Well, to use mysql: then it means the configuration will be also in the
database or I can use mysql only for reporting?
What is the Flash
Hello,
Is it possible to assign templates defined in sip.conf to sip realtime peers?
There was another mail in 2008 which asked the same question but never received
a response.
Thanks,
Alex
--
_
-- Bandwidth and Colocation
Hi
To the best of my knowledge there isn't.
But, if you're using realtime you can create a program to add your
extensions to the database and you can create the concept of templates
within that.
Regards
Ish
On Tue, 2011-09-13 at 12:27 +0200, Alexandru Oniciuc wrote:
Hello,
Is it
Hi All;
Asterisk version is: 1.8.5.0
But I see at the consol the following warning and really I did google but did
not understand if it is bug or related to settings:
[Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just
did sched_add waitid(3429468) for
Kaushal,
Your version of SoX does not have MP3 support. Since you have LAME
installed, use it as a first step to produce an intermediate file
that SoX supports. Then use SoX to convert the intermediate file
to the desired format.
Step 1
--
# lame --decode obd-demo.mp3 obd-demo.wav
input:
hello Virendra,
thx for your response. but after i made clear to the carrier that i
want the dmtf only via rfc2833
and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed.
Kristijan
2011/9/13 virendra bhati virbh...@gmail.com:
Hi
1st check that how many manager is connected into
I'm trying to use Asterisk as a PSTN simulator to run performance tests for
echo cancellation algorithms. I'm using the following configuration:
SIP - Asterisk 1 Asterisk 2 Echo()
Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan
application.
The problem is
I disabled the echo cancelled on the PRI and the same issues are still popping
up:
PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/22 got hangup, cause 16
Anything else I can try?
Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext.
Stephen H. Gerstacker wrote:
Anything else I can try?
Try switchtype=national just for testing.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Hi ,
What was the solution of that problem ? Did provider change the setting at
there end or else ?
On Tue, Sep 13, 2011 at 7:37 PM, Kristijan Vrban
vrban.l...@googlemail.comwrote:
hello Virendra,
thx for your response. but after i made clear to the carrier that i
want the dmtf only via
On 09/10/2011 09:16 PM, Robert Thomas wrote:
Hello List,
I have been trying to configure a sip profile ( peer / friend ) for
each of my customers behind a sip proxy for some time, but I have had no
success, so I would appreciate your help.
Customer - OpenSIPS - Asterisk - PSTN
The opensips
On 09/13/2011 08:56 AM, Gustavo Santos wrote:
I'm trying to use Asterisk as a PSTN simulator to run performance tests
for echo cancellation algorithms. I'm using the following configuration:
SIP - Asterisk 1 Asterisk 2 Echo()
Asterisk 1 and Asterisk 2 are connected using E1.
Ok... this is probably a dumb question but I can't figure out how to set
voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I
pointed it to my asterisk installation, but with IAX I don't have that
option. Is that supposed to work some other way?
Thanks a bunch!
On Mon,
Did you read the “IAX/SIP registration” section (under Authentication) on
voip.ms?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
Sent: Tuesday, September 13, 2011 2:22 PM
To: John Novack
Cc: Asterisk Users Mailing List -
On 12/09/11 09:48 PM, Joseph wrote:
Was NAT problem fixed in 1.8.7 ? I'm using 1.4 but I've tried 1.6 and
1.8.5 and in both cases connection is not working with my provider with
SIP + NAT.
The connection is showing up as registered but the call is not coming IN
(congestion).
Can you define NAT
On 12/09/11 02:21 PM, linux guy wrote:
I'm about to start building my asterisk server and I can't seem to find
anything that discusses the pros and cons of installing the OS (Fedora
15) as console only or GUI, ie install KDE as well.
I see the section you are talking about. It is on the home page if I am not
logged in. I see the Authentication section and the text IAX/SIP
registration, but it doesn't seem to be a link. I am not sure how I can
find the page that has the details about the IAX/SIP registration. I see in
the wiki
I see what you mean. Maybe if you call their support they can tell you what
you need to know. If not, voicepulse is a pretty good provider.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
Sent: Tuesday, September 13, 2011
I was lurking in this conversation and I went to look more carefully
at the voip.ms site. I found sample files at
http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29
Hope that helps.
On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote:
I see the section you are talking about. It
¿How can i could to Send DTMF digits on a current call by scripting
or similar methods?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
http://www.voip-info.org/wiki/view/Asterisk+cmd+SendDTMF
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ezequiel Lovelle
Sent: Tuesday, September 13, 2011 3:37 PM
To: Asterisk Users
Subject: [asterisk-users] Send DTMF
¿How can i
Yup, that part I got. What I am not clear about is how to set up the DID to
go to my URI. When I select manage DIDs and click on the one I want to
change, I see the following options for routing the DID
x SIP/IAX - [main account] IAX2/10 - with my account number
x SIP URI -
That’s what this part of extensions.conf should do:
; inbound context example for your DID numbers, do not add the number 1 in front
[voipms-inbound]
exten = 7863643011,1,Answer() ;your DID
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Sorry if this is an obvious question and perhaps my Google foo isn't
right on this one:
I have calls coming into an Asterisk server that may be using 2
different codecs. I am recording audio in both cases but the
challenge is knowing which codec was negotiated at call setup. I need
to pass the
I'm using them for inbound and outbound on Asterisk and FreeSwitch
Sent from my iPhone
On Sep 13, 2011, at 5:14 PM, Danny Nicholas da...@debsinc.com wrote:
That’s what this part of extensions.conf should do:
; inbound context example for your DID numbers, do not add the number 1 in
front
Sip show channels will give you the active codec. You can get the
information using an AGI or a system command.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Browning
Sent: Tuesday, September 13, 2011
That's what I am hoping to do as well. Could you share some insight on how
you set up the DID on the voip.ms web site to forward to Asterisk using IAX?
In particular I am trying to find out where you set the url / ip address of
your asterisk installation on the voip.ms web site.
Thanks!
On Tue,
Hi
Is there a way to use variables in voicemail.conf.
I want to have an oncall tech system. The tech oncall has his number and email
set in astdb. When a tech call comes in the dial plan checks astdb and sets 2
variable ${oc} for the number and ${ocem} for the email. I can easily dial the
If you change ${ocem} to ${ocem}@default, this will probably work as you
want it to.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelly opal
Sent: Tuesday, September 13, 2011 4:36 PM
To: asterisk-users@lists.digium.com
Subject:
actually Bilal,
the Asteirsk CDR reports are placed on a different Database than the
configurations .. you will need to install asterisk-addons which includes a
module for cdr reporting to MYSQL DB. so you don't have to do the configs from
the DB at all
second.. in regards to the Flash
I would not use voicemail for this.
I would do the following.
Have the call come in.
Ring the on call tech with a dial.
If they don't pickup and press an accept key then.
Answer the call.
Record a message from the caller.
Use a script to e-mail the message to the tech.
(I would
you didn't provide your dialplan for the incoming call context from_poland?
nor registration string?
could be a dial plan problem .. or codec issue.. as long as you register
properly the server has no problem with NAT.. it's a routing or codec issue i
think.
Tarek Sawah
Information
Voip.ms has excellent support if you need it, which many do not.
You log in to your account, then you can change from SIP to IAX, and if you
click on the correct link they will give you your sample with your account
information
You need to set up a registration line in IAX, then a context in
i did do some Asterisk tests on SUN VBOX .. works like a charm but you need to
dedicate some good resources to the virtual box!
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
From:
try to look for N82 nokia mobile devices.. you get the benefits of a Mobile
device with it's phone book and mobility features (games when you are bored :P)
.. and other features.. and the native SIP client works fluently with no
problems at all supporting almost commercial codecs like (G729)..
Ok that makes sense. I will take a look at my set up and see why it is not
registering with voip.ms.
I opened a ticket with voip.ms as well about an hour ago. I do like their
service as well, that is why I want to try and get it working with them.
Thanks John.
On Tue, Sep 13, 2011 at 5:29 PM,
Hi all,
I presume i made a silly mistake while filling a database
But while googling on the results, i came across a lot of messages about
the layout of app_data in case of goto and dial statements.
(mostly about using the old | seperator instead of the , separator.
So i was wondering, is
naren wrote:
Ok that makes sense. I will take a look at my set up and see why it is not
registering with voip.ms http://voip.ms.
Understand that with IAX, voip.ms will not show you as registered.
Your Asterisk should show you as registered from the CLI
CLI iax2 show registry
I made the switch and everything seems to be working. It's hard to tell, since
it never seems to fail for me, but fails once people get in.
A question, though. When we moved the original box to our new office, they
asked if we could support the dms100 setting, which worked. National seems to
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/custom/${TOPMENU})})
im trying to see if a file is available before playing the file
or does anybody have a different idea but not using agi
asterisk 1.6.2.20
thanks
--
On Tue, Sep 13, 2011 at 6:47 PM, Matthew J. Roth mr...@imminc.com wrote:
Kaushal,
Your version of SoX does not have MP3 support. Since you have LAME
installed, use it as a first step to produce an intermediate file
that SoX supports. Then use SoX to convert the intermediate file
to the
On Wed, 14 Sep 2011, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/custom/${TOPMENU})})
im trying to see if a file is available before playing the file
or does anybody have a different idea but not using
On Wed, 14 Sep 2011, Kaushal Shriyan wrote:
Also please let me know the difference between .ulaw and .alaw format
and is there a way i can play this file formats.
alaw = Europe, ulaw = US Japan
Wikipedia has articles on both algorithms if you are interested in the
specifics.
If you
On Wed, Sep 14, 2011 at 6:42 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Sep 2011, Kaushal Shriyan wrote:
Also please let me know the difference between .ulaw and .alaw format and
is there a way i can play this file formats.
alaw = Europe, ulaw = US Japan
Wikipedia has
On Wed, 14 Sep 2011, Kaushal Shriyan wrote:
I have carried out the below steps
[root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.alaw
sox: Output file obd-demo.alaw: using sample rate 8000
size bytes, encoding u-law, 1 channel
Sox v14.x complains about the
On 09/13/2011 07:49 PM, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/custom/${TOPMENU})})
im trying to see if a file is available before playing the file
or does anybody have a different idea but not using
Hi Tzafrir,
On Sat, Sep 10, 2011 at 4:28 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Fri, Sep 09, 2011 at 01:28:28PM -0500, Gerardo Barajas wrote:
There are a lot of reporting tools.
I have used:
Asternic: http://www.asternic.biz/
Non of those are Free (Open Source).
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