On Fri, 16 Jul 2010, bhrugu mehta wrote:
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.
Can't you
On Fri, 16 Jul 2010, Vidura Senadeera wrote:
Dear All,
I am experiance a issue with my IAX clients. I have upgradeed Asterisk to
1.4.28
After then IAX clients are not working and It's not registering even.
Please help.
Put
requirecalltoken=no
in iax.conf for each account.
Gordon
--
В Чтв, 15/07/2010 в 10:44 -0500, Jason Parker пишет:
On 07/15/2010 08:16 AM, Vasiliy G Tolstov wrote:
Hello.
Who can add asterisk16-xmpp module to packages.asterisk.org or build
asterisk with support xmpp and update packages?
Thank You.
This is something we've been considering for a
Hi,
Our guest today is Steven Johnson, President of Ingate Systems. We'll
talk with him about the changing role of Session Border Controllers
and E-SBC and how this hardware facilitates the use of SIP in
difficult conditions, about general SIP security considerations, and
why you might need such
Hello Asterisk-Community,
I'm having an error with my BLF configuration on my asterisk...i've
configured the sip peer like this:
[8250]
type=friend
callerid=Extensión 8250 8250
canreinvite=no
context=pbx9
dtmfmode=rfc2833
host=dynamic
insecure=no
language=es
nat=yes
pickupgroup=
callgroup=
Hello
I just coding a AGI script for billing.
- For external calls, I pass the call directly on a trunk. I do :
Dial(trunk1/extension) - OK !
- For internal calls (shortcode, others users ...) I am
Dial(Local/extens...@context/n)
The problem is that through chan_local.so, I sound as
On 16 Jul 2010, at 09:17, Danny Dias wrote:
[Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766
handle_request_subscribe: SUBSCRIBE failure: unrecognized format:
'multipart/related' pvt: subscribed: 0, stateid: -1, laststate: 0,
dialogver: 0, subscribecont: 'pbx9', subscribeuri: ''
Looks like
On Fri, 16 Jul 2010, Vidura Senadeera wrote:
I am experiance a issue with my IAX clients. I have upgradeed Asterisk
to 1.4.28 After then IAX clients are not working and It's not
registering even.
On Fri, 16 Jul 2010, Gordon Henderson wrote:
Put
requirecalltoken=no
in iax.conf for
yes, actually this scenario is on remote servers. like
SIP/x...@119.18.230.20:5060
SIP/x...@202.68.0.90:5678
audio is ok when dialing without using ip port as below
SIP/XYZ
but when i dial using below dialstring
SIP/x...@202.68.0.90:5678
or
requirecalltoken does not work in the [general] section it must be defined
per peer.
On Fri, Jul 16, 2010 at 5:43 AM, Steve Edwards asterisk@sedwards.comwrote:
On Fri, 16 Jul 2010, Vidura Senadeera wrote:
I am experiance a issue with my IAX clients. I have upgradeed Asterisk
to 1.4.28
On 07/15/2010 08:57 PM, Neeraj Chand wrote:
Hi All,
After getting licences for Skype for asterisk a while ago I finally
got
around to setting up a server with two channels and setting up a bcp
on
the skype end.
The idea behind this is the following:
Users can dial into the PBX, get
On Thu, 15 Jul 2010 12:39:51 -0500, Danny Nicholas
da...@debsinc.com wrote:
This how I would do it
BTW, is it possible to trigger an AGI script right from the first step
and handle the whole IVR logic in an higher-level script language than
what's available in extensions.conf?
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, July 16, 2010 9:28 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Good script to make appointment?
On Thu, 15 Jul
Hello list ?!
Is there anyone that can point me to the documentation please ?
I have added a new table like on
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
With the following values :
`musiconhold` (`name`, `directory`, `application`, `mode`, `digit`,
`sort`,
thank you guys for your responses!
sorry, actually i was not accurate in asking this question, my search is
restricted to soft-phone to use within Black Berry and integrate with
Asterisk.
It seems like the cheapest solution available now that you can integrate
with asterisk and install in Black
On Fri, 16 Jul 2010, Vidura Senadeera wrote:
I am experiance a issue with my IAX clients. I have upgradeed
Asterisk to 1.4.28 After then IAX clients are not working and It's
not registering even.
On Fri, 16 Jul 2010, Gordon Henderson wrote:
Put
requirecalltoken=no
in iax.conf
Based on the info you provided (though wireshark analysis will tell more
about it), I am sure what is happening is that rtp coming back from the
called doesn't know which ip to go to, because asterisk knows two ip
addressses for the same extension due to the way you dialed it, i.e. in
ringgroup
Hi all
A quick question about busy lamps
I have lamps working 'sort-of' on my GXP2000 lamps flash with ringing and
go solid red when call gets answered but stay green when a call is made from
the extension.
Setup is Ext 200, 201, 202, each monitor the other two
when 200 calls 202 - the BLF
It looks like theres no much information out there about using realtime moh
Have you tried making an extension that goes to MusicOnHold(testmoh)
On Fri, Jul 16, 2010 at 8:21 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list ?!
Is there anyone that can point me to the documentation
A very nice feature of another conferencing system that I've used is
that the admin/moderator can press a star code to MUTE ALL OTHER USERS
on the conference.
This is great if you have several people on the call and one of the
people puts the call on hold (and so the music/advertisement/your call
Ok I have a queue that is working perfectly.
The problem is when one of the agents is outside the building on an
external phone line (say a cell phone). My telco hangs up on the call .
I think the telco is hanging up on these calls because there is no CID
attached. (I know my telco wont
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Friday, July 16, 2010 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] (no subject)
Ok I have a
On Thu, Jul 15, 2010 at 10:19:10AM -0700, Steve Casto wrote:
https://issues.asterisk.org/view.php?id=17529
Thanks Tzafrir:
Unclear on how to apply patch, here is what I get:
[root at localhost asterisk-1.4.32]# patch -p1 ../bug17529.diff.txt
can't find file to patch at input line 5
Hi folks,
I have a queue with 5 members, but I need to ring ONLY ONE MEMBER for
every call thats arrive to queue. If another call in incoming, they have
to stay in queue until someone on the queue pickup de call.
Example:
-Queue members-
DAHDI/3
SIP/1000
SIP/1001
SIP/1002
SIP/1003
SIP/1004
A
On Fri, 2010-07-16 at 09:35 -0700, Kyle Kienapfel wrote:
It looks like theres no much information out there about using realtime moh
Have you tried making an extension that goes to MusicOnHold(testmoh)
On Fri, Jul 16, 2010 at 8:21 AM, Jonas Kellens jonas.kell...@telenet.be
wrote:
Hello
Thank you for your input. It seemed like a good approach, but it
confirms that Asterisk does not see the new MusicOnHold-class :
The dialplan :
exten = 60,1,NoOp()
exten = 60,n,MusicOnHold(testmoh)
The CLI :
[Jul 16 19:40:45] -- Executing [...@from-test:2]
Request For Comments on a Feature Suggestion -- just wondering if others
would find this useful.
Frequently, when something really doesn't make sense, I like to bump up
the console logging by editing logger.conf and changing
console = error
to
console =
On 07/16/2010 07:43 PM, Carlos Chavez wrote:
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default '',
`application` varchar(255) collate utf8_unicode_ci NOT NULL
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, July 16, 2010 1:00 PM
To: Asterisk Users Mailing List
Subject: [asterisk-users] RFCFS - reload specified file
Request For Comments
On Friday 16 July 2010 12:59:43 Steve Edwards wrote:
Request For Comments on a Feature Suggestion -- just wondering if others
would find this useful.
Frequently, when something really doesn't make sense, I like to bump up
the console logging by editing logger.conf and changing
On Fri, 2010-07-16 at 20:06 +0200, Jonas Kellens wrote:
On 07/16/2010 07:43 PM, Carlos Chavez wrote:
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default '',
Your solution does not work for me. I've just also added the 'default'
class to my realtime DB.
[Jul 16 20:27:02] -- Called test6
[Jul 16 20:27:02] -- SIP/test6-0014 is ringing
[Jul 16 20:27:04] -- SIP/test6-0014 answered SIP/test2-0013
[Jul 16 20:27:06] -- Started
Hi
Is it possible to receive video calls using Asterisk and then process them
as an IVR ? One of our clients wants to set-up a video IVR system in the US
and we are evaluation possible options.
Also, what is the bandwidth of receiving a video call in US ? What protocols
and codecs are supported
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anita Hall
Sent: Friday, July 16, 2010 1:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Video IVR Asterisk ?
Hi
Is it possible to receive video
Anita,
It's possible to do so. With I6NET
Vxml sofware and some addons. Check out http://www.i6net.com
Adolphe Cher-aime
From my Iphone
On Jul 16, 2010, at 1:38 PM, Anita Hall anita.h...@simmortel.com
wrote:
Hi
Is it possible to receive video calls using Asterisk and then
On Sat, 2010-07-17 at 00:08 +0530, Anita Hall wrote:
Hi
Is it possible to receive video calls using Asterisk and then process
them as an IVR ? One of our clients wants to set-up a video IVR system
in the US and we are evaluation possible options.
Also, what is the bandwidth of receiving
On Fri, Jul 16, 2010 at 2:38 PM, Anita Hall anita.h...@simmortel.com wrote:
Is it possible to receive video calls using Asterisk and then process them
as an IVR ? One of our clients wants to set-up a video IVR system in the US
and we are evaluation possible options.
No, Asterisk does not
Hello Everyone,
I've successfully registered my g729a licenses.
When i try to load the module from asterisk Cli i got the following error
*Error loading module 'codec_g729a.so':
/usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after
reloc:
Try disabling SELinux if you have it enabled (unless of course you need it).
I seem to remember there is certain compilation flags required (position
independent code, -fPIC?) to run with SELinux enabled, may be the
codec_g729a.so is not compiled properly to run under such circumstances?
Moises
Thank you Moises it works perfectly.
On Fri, Jul 16, 2010 at 5:07 PM, Moises Silva moises.si...@gmail.comwrote:
Try disabling SELinux if you have it enabled (unless of course you need
it). I seem to remember there is certain compilation flags required
(position independent code, -fPIC?)
It appears that there's no way to get the return value from a GOSUB into
an AGI script. Is that correct?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
On Friday 16 July 2010 23:35:01 Richard Kenner wrote:
It appears that there's no way to get the return value from a GOSUB into
an AGI script. Is that correct?
No.
GET VARIABLE GOSUB_RETVAL
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig
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