Displaying 20 results from an estimated 10000 matches similar to: "Faking RTP"
2016 Oct 13
2
Openfile Issue
[root at abc asterisk]# lsof -u 50771 | wc -l
0
BTW, I'm using CentOS 6.5
>
> Date: Thu, 13 Oct 2016 10:20:19 -0400
>> From: Dovid Bender <dovid at telecurve.com>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Subject: Re: [asterisk-users] Openfile Issue
>> Message-ID:
2016 Aug 12
2
loosing audio from one end after 5 min.
Hi
Is the keep alive activated on the phone?
On Thu, Aug 11, 2016, 5:36 PM Dovid Bender <dovid at telecurve.com> wrote:
> 1) Does it happen every time at the 5 minute work?
> 2) Have you done a dump on the client side to see if the NAT device is
> dropping the packets?
> 3) Is the phone behind a load balance internet connection and is the RTP
> port changing?
>
>
>
2016 Mar 31
2
Lost outgoing SIP packets
Hi Roel
Just guessing: do you have conntrack enabled?
If not, "modprobe nf_conntrack_netlink" (you can remove it and its dependencies
later)
What are the outputs of
sysctl net.netfilter.nf_conntrack_count
and
sysctl net.netfilter.nf_conntrack_max
when the problem shows up?
cheers
Ethy
On Thu, 31 Mar 2016 12:17:12 +0000
"Dovid Bender" <dovid at telecurve.com>
2019 Aug 01
4
Lightweight ODBC DB
Glenn,
I can't use MySQL as each node currently has MySQL however there is a lot
of data that is stored locally on each box. I may have to take this route
if I can't find something else but that would mean syncing all sorts of
data that does not need to be synced.
On Tue, Jul 30, 2019 at 10:03 PM Glenn Geller (VDOPh) <ggeller at vdo-ph.com>
wrote:
> Hi Dovid,
>
>
2016 May 12
2
maximum call time
Dear Dovid,
thx for the input.
for timer in sip.conf, I used default setting. This is some of the result
for "sip show settings"
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer
2020 Apr 13
1
Multiple real times for same object
On Mon, Apr 13, 2020 at 10:45 AM Joshua C. Colp <jcolp at sangoma.com> wrote:
> On Mon, Apr 13, 2020 at 11:38 AM Dovid Bender <dovid at telecurve.com> wrote:
>
>> Josh,
>>
>> What should Asterisk do if one of the real time methods fail? I have in
>> extconfig.conf
>> musiconhold => curl,http://localhost/moh.php,1
>> musiconhold =>
2017 Dec 27
3
Answered time on channel
It seems that what ever I set in my answer handler does not show up in the
hangup handler. In order to do billing I can't rely on the g option where
the caller hangs up the call. Looks like I can either use h or a hangup
handler along with the shared function.
On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote:
> Don't use an 'h' extension, use
2020 Feb 13
2
Help with FUNC_MATH
John,
That is correct. I am trying to figure out why Asterisk is executing the
set part of the execif, if it's coming back as false.
On Thu, Feb 13, 2020 at 2:10 PM John Kiniston <johnkiniston at gmail.com>
wrote:
> My Apologies Dovid, I think I misunderstood your request.
>
> You don't have the time you need to convert in the format of date string,
> Instead you
2020 Jul 03
0
Exceptionally long queue length queuing
On Fri, Jul 3, 2020 at 3:32 AM Dovid Bender <dovid at telecurve.com> wrote:
>
>
> On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote:
>
>> On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote:
>>
>>> Hi,
>>>
>>> We have a box up and we are starting to see a lot of
2020 Apr 13
2
Multiple real times for same object
Josh,
What should Asterisk do if one of the real time methods fail? I have in
extconfig.conf
musiconhold => curl,http://localhost/moh.php,1
musiconhold => mysql,db-east,asterisk_moh,2
If the first server sends back a 404 it does not go to the second
connection. Shouldn't a 404 be considered a failure and it should then move
over to the next rt engine?
On Thu, Jan 2, 2020 at 7:06 AM
2010 Jan 28
2
rtp.c:883 ast_rtcp_read: RTCP Read too short
Hi:
I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?:
?
-- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack
??? --
2017 Sep 19
0
AST-2017-008: RTP/RTCP information leak
Asterisk Project Security Advisory - AST-2017-008
Product Asterisk
Summary RTP/RTCP information leak
Nature of Advisory Unauthorized data disclosure
Susceptibility Remote Unauthenticated Sessions
Severity Critical
2020 Feb 13
0
Help with FUNC_MATH
My Apologies Dovid, I think I misunderstood your request.
You don't have the time you need to convert in the format of date string,
Instead you have your users entering via DTMF when they want something to
happen?
On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender <dovid at telecurve.com> wrote:
> John,
>
> From looking at the wiki won't STRFIME just give me what I need based
2020 Jul 03
2
Exceptionally long queue length queuing
On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote:
> On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote:
>
>> Hi,
>>
>> We have a box up and we are starting to see a lot of "Exceptionally long
>> queue length queuing" in the logs. From all the research so far it seems
>> like this leads to
2020 Oct 23
0
chan_sip and matching the RTP source
All,
I am stuck with a specific install using chan_sip and Asterisk 11.25.3. We
have nat=no which from what I understand means that Asterisk will go by
whatever it see's in the SDP and not look at the source IP+port from where
the traffic is coming from. We have a call flow where we send a carrier a
call and they specify an IP and port in their SDP in a 183 (e.g.
100.100.100.100:36070). As we
2016 Mar 31
2
Lost outgoing SIP packets
Dovid Bender writes:
> Just guessing I would verify that the out of : iptables -L -nv
> Shows no dropped packets, try disabling selinux as well as look at the
> limits of the asterisk pid (cat /proc/<Asterisk PID>/limits). I know the
> defualt for rhel is 1024 which was never enough for us.
Thanks for the hints. Selinux is disabled, there is no outgoing firewall
(anymore)
2020 Oct 29
0
Expert to work on load issue
Anyone have any other ideas?
On Tue, Oct 27, 2020 at 1:27 PM Dovid Bender <dovid at telecurve.com> wrote:
> Jon,
>
> We are only using FastAgi. On the second system (running Asterisk 16)
> there are no agi's running (just some bash scripts on call hangup). I did
> add some hackey code (netstat -nua | grep -v 'udp 0 0' | grep
> -v udp6 | grep -v
2018 Jun 27
2
Asterisk crashing on AAAA lookup
On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett <rmudgett at digium.com>
wrote:
>
>
> On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender <dovid at telecurve.com> wrote:
>
>> I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so
>> often asterisk crashes and then restarts. I am not seeing any core dumps on
>> the box. The only I thing I see
2020 Oct 27
2
Expert to work on load issue
Jon,
We are only using FastAgi. On the second system (running Asterisk 16) there
are no agi's running (just some bash scripts on call hangup). I did add
some hackey code (netstat -nua | grep -v 'udp 0 0' | grep -v
udp6 | grep -v ' 0 0.0.0.0' | grep udp) to my bash script to check out the
packet queue (with the help of
2019 Apr 19
2
Forking AGI or GoSub
In PHP something like:
$pid = pcntl_fork();
if ($pid != 0) {
// we are the parent
// do parent stuff
exit;
}
// we are the child, detatch from terminal
$sid = posix_setsid();
if ($sid < 0) {
die;
}
// do child stuff
On 04/19/2019 02:00 PM, Mark Wiater wrote:
> On 4/19/2019 1:49 PM, Dovid Bender wrote:
>> Mark,
>>
>> I am using PHP agi and when forking