is sent from the server to the phone
as '0x0'. The same has happened with G729 codec.
> Let me know if you need the full trace or anything else from my side.
> I should also mention that this is Asterisk version 1.8.12.1
> Thank you
> Harel
> --
>
On
the phones? Is this something that can be
fixed on the Asterisk side?
Thank you,
Harel
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is expecting to get its RTP from.
KR
Harel
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Hello All,
For a development of an auto-dialler I need to make a script that will
SUBSCRIBE to the status of certain SIP channels and act upon the changes of
their state.
How can I subscribe over AMI (what 'Action:' do I need to use) and what
response should I look for?
Many Thanks,
Harel
(including the move vs copy)
so my problem is not there.
My main objective is to initiate a call as soon as an agent becomes available
(also using predictive algorithms) and I don't want to waste time on calling
one side and only when answered calling the other.
Harel
e ringback tone until 2nd party
answers)?
Thank you
Harel
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on the [peer] level thus
providing more valuable tools to solve NAT issues.
Harel
Original Message *
Hi,
normally, Asterisk handles RTP IP addresses in SDP correctly, if you have
specified
- that NAT traversal is enabled for all peers (e.g. nat=force_rport,comedia)
- your local
a
professional security box.
Asterisk is 13.6.0
I can't, and don't want to, touch user-side equipment which is normally some
kind of voip phone behind a standard home VDSL router.
Any ideas how can I transmit the correct IP address in SDP to UAs on different
networks?
Many thanks,
Harel
that the hangup()
is never executed but it’s my habit to put one in such small modules to avoid
nasty auto-fallthrough when you don’t expect it…
Kind Regards,
Harel Cohen
Director
Mayorcom Limited
Mobile:+350 58009379
Office: +350 2005
<mailto:ha...@mayorcom.com> ha...@mayorcom.com
Support Level
res_fax.so Generic FAX Applications 1
Running core
res_fax_spandsp.so Spandsp G.711 and T.38 FAX Technologies 0
Running extended
2 modules loaded
Thank you for your assistan
No one could assist?
Could someone please tell me on which repository I can find Gmime22-devel
for 64-bit Centos6.5?
Is gmime-devel good or do I need to have gmime22-devel?
What will happen if I don't install gmime22?
Thank you...
Harel
Message: 3
Date: Mon, 6 Jul 2015 02:53:51 +0200
From: Harel
Hello list,
I'm trying to install gmime22 package which is one of the packages reported
as required by ./contrib/scripts/install_prereq test.
Whatever I do I'm getting to a dead end.
On the regular yum repositories that I use (centos, epel, rpmforge,
asterisk, digium) it is not found.
I've found
this 'OK' RTP packets are sent to 172.24.100.2 instead of
remote_public_IP
Thank you,
Harel
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Attribute (a): sendrecv
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): silenceSupp:off - - - -
Media Attribute (a): ptime:20
Media Attribute (a): maxptime:90
Thank you,
Harel
for your suggestion
though...
Any other thoughts are welcome.
Kind Regards,
Harel Cohen
-Original Message-
Date: Mon, 12 Dec 2011 13:41:31 -0700
From: Mike Diehl mdi...@diehlnet.com
Subject: Re: [asterisk-users] Populate CDR issues
To: asterisk-users@lists.digium.com
Cc: Harel Cohen
,
otherwise I'll simply follow ASTERISK-18875. My problem with this issue is that
it is defined as low importance which means that it will probably take long to
handle if at all...
Harel
**
Message: 4
Date: Tue, 6 Dec 2011 07:29:54 -0600
From
or '-') and NOT with the values above.
Am I doing something wrong or is there a different way to populate CDR's with
info from called channel (leg B)?
Thank you for your replies...
Harel
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or '-') and NOT with the values above.
Am I doing something wrong or is there a different way to populate CDR's with
info from called channel (leg B)?
Thank you for your replies...
Harel
This electronic message and any files transmitted
also mess around with renaming files using System() however I was hoping there
is a straight forward way rather than work-around.
Thank you...
Harel
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New
the call upon pick up.
I can use any version of Asterisk as required.
Any opinions and ideas would be appreciated.
Kind Regards,
Harel
This electronic message and any files transmitted with it are confidential and
intended solely for the use of the individual
Hi All,
I would like to separate the media traffic from the signalling.
Can Asterisk send and receive media (rtp) traffic from a secondary network
interface?
Thanks,
Harel
This electronic message and any files transmitted with it are confidential and
intended
Hi All,
I would like to separate the media traffic from the signalling.
Can Asterisk send and receive media (rtp) traffic from a secondary network
interface?
Thanks,
Harel
This electronic message and any files transmitted with it are confidential and
intended
Please ignore this message (wrong subject by mistake). Please see message with
subject 2nd network interface for RTP/media
Thanks
Harel
--
Message: 2
Date: Mon, 1 Nov 2010 12:52:16 +0100
From: Harel Cohen ha...@easycall.gi
Subject: [asterisk-users] MoH and stuch
Hi all.
Is there a free, or at least non-expensive, solution that can convert g729A
--g729B (with VAD)? The no-support for g729B on Asterisk gives me a BIG
headache…
Thanks,
Harel
This electronic message and any files transmitted with it are confidential
-versa. Could you (or someone) please take another
look to locate the correct file?
Thanks
Harel
--
Message: 4
Date: Wed, 04 Aug 2010 15:20:05 +0200
From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Subject: Re: [asterisk-users] mapping of disconnect
to the relevant file of code where these mappings are done?
Before reporting a bug I would like to confirm whether this issue has been
addressed on newer releases.
Thanks,
Harel
--
On Tuesday 03 August 2010 06:21:23 Philipp von Klitzing wrote:
Is there a way
dial plan will look for
alternative termination in the event of network error (e.g. reason 3 or 21
which is resulting call status “CONGESTION”) but will not do so for all normal
terminations (16, Normal Termnation, 17 Busy, 18 No Answer).
Thanks,
Harel
times or certain intervals?
Thanks,
Harel
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the phone in one room and
picking up in another room without disconnecting the line. Make a small test to
verify this and if this is the case you will need to discuss this with your
PSTN provider.
Harel
Date: Thu, 8 Jul 2010 12:01:40 -0500
From: Daniel - Asterisk earohua...@gmail.com
Subject
extension (Internal, 22, 1) exited non-zero on 'SIP/309-00a5'
If I use the ‘g’ option in my Dial() both Noop will be run only if the called
party hang-up first. I need a simple continuation in the dial plan regardless
of who disconnected the call.
Thanks in advance
Harel
ast_readaudio_callback: Failed to write frame).
Tiago:
There is no Dial() option to simply continue dial-plan after Dial(). See above
regarding g option.
Can anyone think of a way to play IVR after conversation initiated by Dial()
terminates?
Harel
--
Message: 9
would consume fewer resources (put aside other pro's con's)?
Is there any better way of implementing this?
Would 'hints' help me out here? If yes, I would appreciate a detailed
explanation how to use it.
Thanks in advance,
Harel
on this
issue if needed).
Thanks in advance for any ideas provided,
Harel
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) or do I still have
the ability to use this G729 codec for other call which requires transcoding?
Thank you,
Harel Cohen
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