find the inline comment...
On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
The dialplan is very simple. When the call comes in, we hand the call
over to adhearsion.
This is how the dialplan looks:
;group 0 will be used for incoming calls
EXOIN = DAHDI/g0
;group 11 for outgoing
EXOOUT =
Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that
you will receive in that ,
also read this for better implementation.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
regards
Dhaval
On Fri, Jul 29, 2011 at 11:58 AM, Nikhil d.nik...@cem-solutions.net
Hi!
I need help regarding the following problem:
when I receive a phone call to the PBX from the number 01234567890
rings the number 100, get up the phone, I transfer (assisted) to the number 100.
When the 100 number rings, the display shows the number of those who
transferred the call and not
That`s the normal behavior of assisted transfers. Try a blind/non-assisted
transfer, that should show the original callerid.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 2:56 AM
To:
Thanks for the reply!
I've tried and works, but isn't possible with the transfer assisted?
thanks
From: Mike
Sent: Friday, July 29, 2011 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] call forwarding number from outside.
That`s the
Hi,
I haven't write any How to on it but below are some step by step
instructions to run Asterisk on windows,
1-Install Cygwin.
2-Install build essentials in Cygwin.
3-Download Asterisk source (I used 1.4.x) and unzip it using tar (You may
need to install tar manually as it is missing in some
Did you tried to execute Set(CALLERID(num)=you-required-callerid)?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Friday, July 29, 2011 1:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
One more thing previously there was a project named as AstWin which was
maintaining asterisk's port to windows and providing an installable package
of Asterisk for windows. I am not aware about current state of project but,
I have installation package of Asterisk for windows version 1.2.
On Thursday 28 Jul 2011, Gilles wrote:
On Thu, 28 Jul 2011 12:04:38 +0500, Faisal Hanif fai...@vopium.com
wrote:
I have tried asterisk on windows XP using Cygwin and it worked fine.
Would you mind explaining how to do this?
I hate to sound patronising but, if you need to ask how to install
Thank you Terry, CallWithUs is what I am looking for, the most feature
rich VoIP service!!!
I hope it will not be difficult for me to have it working with Asterisk and
OpenBTS (It's worth to see what OpenBTS is)
On Thu, Jul 28, 2011 at 12:48 PM, Terry Brummell te...@brummell.net wrote:
Yes,
Hello,
Asterisk seems to try to authenticate incoming INVITE based on the [section]
in sip.conf and not the username specified.
I just removed the insecure option from my sip.conf requesting every
connection to be authenticated. I added the match_auth_username=yes in the
[general] section for
Voip.ms actually offers more features. Depends on your needs. I use
both as long distance carriers. My DID's are from Voip.ms.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A.H. Jos
Sent: Friday, July 29, 2011 4:53 AM
To:
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am
seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam
/usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl
Hi List,
I want to use these features but nothing was found after googling . please
give me some examples
Asterisk CLI prompt
Changing the CLI Prompt
The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable
that
you set from the Unix shell before starting the Asterisk CLI (not
Use This Information.
You can customize the prompt a bit, if the default prompt is too dull for
you. First add these lines to */etc/asterisk/extensions.conf* in the
[globals] section:
${ENV(UNIX)}
${ENV(ASTERISK_PROMPT)}
Then in */etc/profile* on the Asterisk server, set the ASTERISK_PROMPT
On Friday 29 Jul 2011, virendra bhati wrote:
Hi List,
I want to use these features but nothing was found after googling . please
give me some examples
Asterisk CLI prompt
Changing the CLI Prompt
The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable
that
you set from
- Original Message -
What you are wanting to do is, in essence, like teaching a gerbil to
bark.
It's an extraordinary effort to go to when you can get puppies
anywhere; and
at the end of the day, it's still not, and never will be, a proper
dog.
+1 I could not have said it better
The issue with assisted transfer is that the assisting transferer is a
second call
Outside - A
A answers
A calls B to tell them they have a call (call #2 with ID of A
A transfers Outside but the ID stays A
Blind Transfer
Outside - A
A answers
A blind transfers to B (1 call - keeps ID.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, July 29, 2011 9:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: jim.smith...@debsinc.com
Subject: Re:
On 07/29/2011 09:12 AM, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, July 29, 2011 9:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
HI Eric, Nikhil,
Thanks a lot for the responses. Bear with me a little as I'm very new to
asterisk.
I reproduced the problem using standard dialplan. The following are the
configuration files:
*chan_dahdi.conf*
*[trunkgroups]
[channels]
language=en
nationalprefix=+91
pridialplan=national ; or
On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and
am seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for
-lpam
/usr/bin/ld: skipping incompatible
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Friday, July 29, 2011 8:49 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.
On
On 07/29/2011 10:41 AM, Danny Nicholas wrote:
snip
Couple of questions -
This magic trick is contained in app_dial?
Functionality is inherent to 1.8/10.X structure so we can't re-invent this
in our old 1.4/1.6 installs?
No, it's core functionality, implemented in the channel drivers and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Friday, July 29, 2011 9:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.
On
So I can't do anything?
--
From: Kevin P. Fleming kpflem...@digium.com
Sent: Friday, July 29, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.
On 07/29/2011 10:41 AM, Danny
Upgrade to 1.8/10.0
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 10:04 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from
Thank you Dave.
--
Thanks, Phil
- Original Message -
On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6
X86_64 and
am seeing the following when running the make:
/usr/bin/ld: skipping incompatible
Hi team,
Is it possible to capture dtmf input once call is patched between a-party and
b-party? Also on dtmf input issue hangup request to b-party with out
disconnecting A-party.
How is this scenario implemented in dialplan?
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
--
ok I'll do it Monday, and how you handle it with the version 1.10?
Thanks
--
From: Danny Nicholas da...@debsinc.com
Sent: Friday, July 29, 2011 5:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ishwar Sridharan
Sent: Friday, July 29, 2011 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capturing
Yep. Look the dtails of option of Dial command and features.conf.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinod
Dharashive
Sent: Friday, July 29, 2011 8:51 PM
To: asterisk-users@lists.digium.com
HI Eric,
is a mobile number in India, and the call id rejected by ending the
call from the mobile.
BTW, why is the mail going to asterisk-users-bounces?
--
Thanks,
Ishwar.
On Fri, Jul 29, 2011 at 9:34 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From:
Hello,
We enable pri intense debug with the standard asterisk PRI dialplan,
collected the logs and you can find the logs attached to the mail.
After the call was made, the called party cut the call, and asterisk doesn't
seem to recognise the event.
I can't make much sense of the logs given my
We enable pri intense debug with the standard asterisk PRI dialplan,
collected the logs and you can find the logs attached to the mail.
After the call was made, the called party cut the call, and asterisk
doesn't seem to recognise the event.
I can't make much sense of the logs given my
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/jira/browse/ASTERISK-18103
What is the general time to fix this? I think a similar thing is also noted
in 1.8x install. Is it not going to be taken care of because it's 1.6x ?
Thanks
--
On 11-07-29 06:12 PM, Bruce B wrote:
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/jira/browse/ASTERISK-18103
What is the general time to fix this? I think a similar thing is also noted
in 1.8x install. Is it not going to be taken care of because it's 1.6x ?
On 07/29/2011 06:20 PM, Paul Belanger wrote:
On 11-07-29 06:12 PM, Bruce B wrote:
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/jira/browse/ASTERISK-18103
What is the general time to fix this? I think a similar thing is also
noted
in 1.8x install. Is it not going
I've used the manager interface to make calls successfully, now I'd like
a look at some of he other ways it can be used.
I've seen references to its use to perform call cut off and rate CDRs.
Is anyone aware of a reference or tutorial I could look at?
Bruce Ferrell
--
I think this should be a quick fix since it's rendering the latest stable
version useless and making the impression that it was released just to break
things and force people onto 1.8x. Just a thought...no blame game. But
really something like this should be tackled quickly. No point to break
40 matches
Mail list logo
