Thanks for reply,
I had check it. in auto dialer whenever dial the number there is no voice to
get agent. dialer will dial the number asterisk not getting voice like
swo,NA. how we can get the voice in there.
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE |
Friends,
While working with the manager interface, I noticed that an originate action to
a non-existing extension had a strange behaviour. Instead of generating an
error, which would happen in most VoIP channels and Dahdi, Asterisk started
looking for extension s as a fallback.
For as long
18 sep 2011 kl. 22:23 skrev Catalin S.:
Hello Eric,
Is about outgoing calls from multiple devices with the same username at aprox
same time. The overwritten is for incomming calls. I want to prevent using
the same account in multiple devices at same time. The solution with IP will
not
On Tue, 2011-09-20 at 11:26 +0200, Olle E. Johansson wrote:
18 sep 2011 kl. 22:23 skrev Catalin S.:
Hello Eric,
Is about outgoing calls from multiple devices with the same username at
aprox same time. The overwritten is for incomming calls. I want to prevent
using the same account
19 sep 2011 kl. 01:51 skrev Zeeshan A Zakaria:
This DTMF problem has always been there and there is no real solution for it,
other than using those expensive PBX systems like that from Avaya, Cisco,
etc. This problem happens when you are sending inband DTMF tones. Via
softphone you are
Thanks for your reply, but it was an issue with upstream provider. The
number got stuck in the middle of being ported.
On 09/20/2011 01:34 AM, C F wrote:
Without your dialplan there isnt much that can be done to help.
Can you please post your relevant dialplans?
Whats voip1 and voip2?
When you
Please check offline message
Regards
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Tue, Sep 20, 2011 at 2:47 AM, mahesh katta maheshka...@flexydial.comwrote:
Thanks for reply,
I had check it. in auto dialer whenever dial the number there is no voice
to get agent. dialer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E.
Johansson
Sent: Tuesday, September 20, 2011 4:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Fixing an old bug
Hi List,
I currently have a asterisk server running used for dialing-out for IDD but I
want to Put a pincode wherein only users with the right pin code will be
allowed to dial IDD. Any sample dialplan you can suggest pls?
Thanks,
Malvin--
That's what the DISA function is for.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito
Sent: Tuesday, September 20, 2011 8:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Add PinCode on my dialplan
Or authenticate aplication.
If you want use a database with a user and pin table, so each user have a pin
asigned, you can look a func_odbc function.
Regards
- Original Message -
From: Danny Nicholas
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Tuesday,
+1 bakko
Using DISA might open a hole you don't want to have
asterisk -rx core show application disa
-= Info about application 'DISA' =-
[Synopsis]
DISA (Direct Inward System Access)
[Description]
DISA(numeric passcode[|context]) or DISA(filename)
The DISA, Direct Inward
20 sep 2011 kl. 15:34 skrev Danny Nicholas:
Just my .02 - fix Originate since the Original Asterisk book, page 125
paragraph 1 says s = start. If s is not really start, I'm going to
scrap my 3+ years of dialplan writing and change all of my simple dialplans
to read exten= start,1,blah
20 sep 2011 kl. 15:34 skrev Danny Nicholas:
Just my .02 - fix Originate since the Original Asterisk book, page 125
paragraph 1 says s = start. If s is not really start, I'm going to
In the first edition, page 82, it actually says When a call enter a context
without a specific destination
Sorry if this question already asked.
I'm implementing Voip with asterisk and grandstream gxw4108, according
from the manual, for connecting with PSTN I must configure one SIP
account and assign that for dialing the PSTN so in my sip.conf I
configure SIP account(extension) :
[1401]
type=friend
Hi all,
I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to
add 2 new modules : jabber and chan_gtalk.
How to add these modules and not change anything of configuration existed
Asterisk ?
Best regards,
Ryan.
--
Just do make menuselect
Then
Make make install
As long as you don't do any of the other steps after make install, no
configuration files should be updated.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tran quoc tuan
Sent:
Thank Danny Nicholas for your reply ,
It means I may re-make menuselect in Asterisk version 1.4.36 to add 2 new
modules : jabber and chan_gtalk ? Can version Asterisk 1.4.36 support these
module : jabber and chan_gtalk ?
Thank in advance for all helps!
B.R
Ryan.
On Tue, Sep 20, 2011 at 10:39
To the best of my knowledge, these modules have been supported at least
since 1.4.22.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tran quoc tuan
Sent: Tuesday, September 20, 2011 10:46 AM
To: Asterisk Users Mailing List -
Thank you very much. I will try to install these modules.
B.R.
Ryan.
On Tue, Sep 20, 2011 at 10:52 PM, Danny Nicholas da...@debsinc.com wrote:
To the best of my knowledge, these modules have been supported at least
since 1.4.22.
** **
*From:* asterisk-users-boun...@lists.digium.com
What is the difference between using mISDN for BRI and using Dahdi without
mISDN?
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
At 07:09 AM 9/20/2011, you wrote:
Using start makes your dialplans much easier to read :-) and makes
them more secure as no app will end up there by accident, which may
happen in your current systems.
When I went and read version 3 it seemed to indicate that start has
no actual meaning and I
If I have a 4 port Digium FXS card and a single port PRI card on the
same asterisk box, is it expected that I'd be able to plug a fax machine
into the analog FXS port and have no problems sending or receiving
faxes? Our connection to the Telco is on the PRI obviously.
I don't recall the
On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett adamli...@plexicomm.netwrote:
If I have a 4 port Digium FXS card and a single port PRI card on the same
asterisk box, is it expected that I'd be able to plug a fax machine into the
analog FXS port and have no problems sending or receiving faxes?
Am 20.09.2011 19:47, schrieb Gopal krishnan:
What is the difference between using mISDN for BRI and using Dahdi
mISDN was at 1st done for ISDN Services and channel driver as I know. It
supported like call routing (switch based, not your side on the pbx level).
without mISDN?
you can use
I am having a problem with one of my sites where they are not receiving the
voicemail to email. I've done a lot of troubleshooting and can't find the
issue. It would be helpful if there was a log I could look at so that I could
see perhaps where the email is being rejected. Does anyone know of
On 20/09/11 06:53 PM, Kevin Oravits wrote:
I am having a problem with one of my sites where they are not receiving
the voicemail to email. I’ve done a lot of troubleshooting and can’t
find the issue. It would be helpful if there was a log I could look at
so that I could see perhaps where the
On 20/09/11 09:34 AM, Danny Nicholas wrote:
Just my .02 - fix Originate since the Original Asterisk book, page 125
paragraph 1 says s = start. If s is not really start, I'm going to
scrap my 3+ years of dialplan writing and change all of my simple dialplans
to read exten= start,1,blah instead
On 20/09/11 03:37 PM, Ira wrote:
At 07:09 AM 9/20/2011, you wrote:
Using start makes your dialplans much easier to read :-) and makes
them more secure as no app will end up there by accident, which may
happen in your current systems.
When I went and read version 3 it seemed to indicate that
On 09/20/11 17:53, Kevin Oravits wrote:
I am having a problem with one of my sites where they are not receiving
the voicemail to email. I’ve done a lot of troubleshooting and can’t
find the issue. It would be helpful if there was a log I could look at
so that I could see perhaps where the email
Is anyone can help me with this ? I'm really desperate.
Thx in ad.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ikka - Mitra
Kreasindo
Sent: Wednesday, September 14, 2011 5:02 PM
To: 'Asterisk Users Mailing List -
Tarek,
Thank you for your response. I am going with the load balancing idea.
Claude
-- next part --
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20110917/9a831
c42/attachment-0001.htm
hi:
please check the chan_gtalk.conf, add your account info.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Tue, 20 Sep 2011 22:37:00 +0700
From: tuant...@gmail.com
To: asterisk-users@lists.digium.com
On Tue, 20 Sep 2011, Kevin Oravits wrote:
I am having a problem with one of my sites where they are not receiving
the voicemail to email.
They are not receiving or you are not sending?
Unless you've changed 'mailcmd' in voicemail.conf, Asterisk will execute
'/usr/sbin/sendmail -t' to send
I am not real familiar with the size of MixMonitor parameters, but just
looking at the output, I would suggest you change the logic to call a
script with a single argument.
something like this,
MixMonitor(${FILENAME},bW(2),/usr/local/bin/convert_to_mp3 ^{FILENAME})
---
On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett adamli...@plexicomm.net
wrote:
If I have a 4 port Digium FXS card and a single port PRI card on the same
asterisk box, is it expected that I'd be able to plug a fax machine into the
analog FXS port and have no problems sending or receiving faxes? Our
On Tue, 2011-09-20 at 20:57 -0500, Don Kelly wrote:
On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett
adamli...@plexicomm.net wrote:
If I have a 4 port Digium FXS card and a single port PRI card on the
same asterisk box, is it expected that I'd be able to plug a fax
machine into the analog FXS
Something like this should work:
exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)
exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
Just to help with troubleshooting you could try to reproduce the same problem
with a different set of SIP endpoints. Setup a soft phone as the destination
and see if the problem occurs there. That way you can eliminate the handset as
a potential problem.
On Sep 15, 2011, at 10:31 AM,
I have no solution, but my head hurts thinking about listening to separate
calls simultaneously in each ear.
On Sep 9, 2011, at 9:22 AM, fhirschberg wrote:
Hi list!
I'm using the latest Asterisk 1.8.6.0 cross compiled for an i.MX27 board
and it works really good.
But I need a feature and
DISA and DB based Auth could be an overkill.
Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
if Voicemail PIN are required to be used use application
+1 Dale
Alternatively I'd troubles using the MixMonitor() command execution, so what
I did is used System(my commands here) just after the StopMixMonitor().
Using StopMixMonitor() is always recommended to guarantee save the recorded
file and using any commands via System() is easy.
On Wed, Sep
Thanks. ?If I want to use unique PIN for every user that dials out how
can I implement it using Authenticate app?
Regards,
Malvin
On 9/21/2011 12:42 PM, Sam Govind wrote:
DISA and DB based Auth could be an overkill.
Kyle showed the very simplistic dial plan if Dial-out pin is common
for the
Couldn't help LOL on Kyle's remarks. But it could be two users listening to
two different streams/calls. Obviously both can't share single mic on their
call(if they ever need it).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
44 matches
Mail list logo
