Tue, Nov 14, 2017, at 09:14 AM, Harel Cohen wrote:
> > Hello,
> > I have a problem where on an outgoing call a Grandstream phone
> > (GXP2130) closes the incoming voice stream about 1 second into the
> > call (the remote party hears the Grandstream, the Grandstream doesn't
Hello,
I have a problem where on an outgoing call a Grandstream phone (GXP2130)
closes the incoming voice stream about 1 second into the call (the remote
party hears the Grandstream, the Grandstream doesn't hear thr remote
party). I have verified with logs and traces that this is not a NAT issue
Hi,
Is the Sophos a home router or professional one? In many cases what home
router does by default needs to be configured manually on professional one.
E.G. a home router will allow outgoing sessions and create a return path by
default where professional one won't.
Two things I would look for:
1.
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
--
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
Hello List,
I need your advise please.
I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP
UA (not Asterisk), both are behind NAT. That remote peer is configured with
nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP
address which is declared in
Hello List,
I need your advise please.
I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP
UA (not Asterisk), both are behind NAT. That remote peer is configured with
nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP
address which is declared in
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
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harel Cohen
Sent: Tuesday, December 06, 2011 3:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Populate CDR issues
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just in case
someone who might be able to help missed my previous post.
Thank You...
*
Hello list,
I'm trying to populate my CDR logs
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel.
Hello List,
I have few installations out there based on 1.6.1 or above.
I'm trying to play different voice mail messages based on certain criteria's.
For example, I want during office hours to play (in short): we are not
available to take your call, please leave a message, during off-hours and
Hi All,
I would like to implement a call-back option when called user is busy.
Consider this scenario:
1. A caller is calling a number which is busy on another call.
2. The system will prompt the caller (press 3 to be called back etc.) to be
called back when called user is available.
3. Caller
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 and
Sorry for the late response.
Philipp,
I've checked the file below and also the suggested voip-info link. None of
those describe how or why Asterisk assumed that 402 should be mapped to NORMAL
TERMINATION status. Both places refer to how Asterisk status should be mapped
to SIP cause and not
Tilghman, thank you for your reply.
The mapping in RFC 3398 is logically correct therefore I do not need to submit
a suggestion to its editor.
The mapping in Asterisk 1.4.24 is the problem:
402 Payment Required is mapped to 16 Normal termination instead of 21 Call
Rejected.
Could you direct me
Hi All,
Is there a way to change the mappings of disconnect reasons to certain SIP
messages? E.G. I need to change the mapping for SIP 402 “Payment Required” from
16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined
in RFC 3398. For me this is a big issue because my
Hi all,
Can the Asterisk do “things” not during a call? For example I would like to
change my dial plan during certain hours\dates or I would like to check some
information in the astdb (e.g. counters of al sort) and handle it as required
and so on. All of this is not call-related therefore I
Hi Elder,
I would first check the behaviour of your PSTN lines (i.e. nothing to do with
Asterisk). In many places PSTN companies allow between 30 to 90 seconds of
connection to remain open even if the -called- party, NOT the calling party,
has hung-up. Normally this is to allow putting down the
Hi All,
I’m trying to do “things” after my Dial application terminates (e.g. play IVR
to called party, calling party, etc.). I’m trying to use the local channel for
this purpose but so far with no success. I’m using 1.6.1.18 and this is my
extensions.conf:
[Internal]
exten =
A Zakaria
--
www.ilovetovoip.com
On 2010-06-22 6:23 AM, Tiago Geada tiago.ge...@gmail.com wrote:
Hi,
After a Dial, the call is hung up. It doesn't carry on with dialplan unless
you specify the appropriate dial option.
Check wiki voip-info for cmd Dial, I think the option is g
2010/6/22 Harel
Hi all,
Could someone please tell me what is the relative cost in using conf files
oppose to the astdb? Basically I need to match a name to a phone number in
order to have all users registered by name and not by number (which I
understood is not a good practice). I have 2000 users and a complex
Hi all,
How can I implement a full-featured Call-Waiting behavior on the Asterisk level
(e.g. I don't want to relay on end-equipment capabilities)?
I found it very strange that such a basic feature is not built-in in Asterisk
(and I've googled a lot in search for this).
Here is what I need:
) 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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