<1484156556.3940360.844569720.0a814...@webmail.messagingengine.com
> >
> Content-Type: text/plain; charset="utf-8"
>
> On Wed, Jan 11, 2017, at 12:59 PM, Ahmed Munir wrote:
> > Thanks.
> >
> > But I was not able to find the records in 'ps_contacts' table. As per
.6368d...@webmail.messagingengine.com
> >
> Content-Type: text/plain; charset="utf-8"
>
> On Tue, Jan 10, 2017, at 10:11 AM, Ahmed Munir wrote:
> > Hi,
> >
> > I would like to know how to check PJSIP status for endpoints at DB level
> > (real
.8 cert 4
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
="utf-8"
>
> 50771 is the PID. I am talking about the user. for instances if running as
> root (which you should never do) then:
> lsof -u root | wc -l
>
> On Thu, Oct 13, 2016 at 1:31 PM, Ahmed Munir
> wrote:
>
> >
> > [root@abc aster
init script:
>> ulimit -s unlimited
>> ulimit -n 65535
>> ulimit -Hn 65535
>> ulimit -u 65535
>> ulimit -Hu 65535
>>
>>
>>
>> On Thu, Oct 13, 2016 at 9:59 AM, Ahmed Munir
>> wrote:
>>
>> >
>> > See below;
>> &g
conf as well as add to the Asterisk init script:
>> ulimit -s unlimited
>> ulimit -n 65535
>> ulimit -Hn 65535
>> ulimit -u 65535
>> ulimit -Hu 65535
>>
>>
>>
>> On Thu, Oct 13, 2016 at 9:59 AM, Ahmed Munir
>> wrote:
>>
>> >
Content-Type: text/plain; charset="utf-8"
>
> What do you get when you do:
> cat /proc//limits ?
>
> On Thu, Oct 13, 2016 at 9:30 AM, Ahmed Munir
> wrote:
>
> > Hi all,
> >
> > Now a days getting openfile issues on asterisk quite often even setting
Further added, I'm using CentOS 6.5 as OS.
Please advise what changes required for permanently fixing this random
issue.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
for your help.
On Thu, Sep 29, 2016 at 12:18 PM, Ahmed Munir
wrote:
> Hi Guys,
>
> Even though enabling Asterisk debug (setting to 9), getting same message
> and not providing enough logs;
>
> DEBUG[10801][C-]: cdr_radius.c:208 radius_log: Unable to create
> RADIUS
ildreq.c
> >
> > You'll see many instances where it returns ERROR_RC. You are almost
> > certainly running into one of these. I ended up putting in print debug
> into
> > that file and recompiling. I think in my case it was as simple as a
> > hostname not resolving
from.nl>
> Content-Type: text/plain; charset=us-ascii
>
> Hello Ahmed,
>
> On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote:
> > Hi,
> >
> > I've recently setup Asterisk with Radius CDR by following the document:
> > https://wiki.asterisk.org
aptive ODBC
cdr-custom
csv
radius
Please advise if I may missed any steps.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13
obser
In sip.conf the parameter I've enabled/uncommented for realtime are only
'rtcachefriends=yes' and rest of the realtime parameters are commented (set
as default).
Please advise, what I'm may missed out.
--
participants.
Any ideas which may causing this issue? As Asterisk version I'm using is
11.2.1. Is it a bug? Please advise.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -
ercome this issue.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.a
s Mailing List - Non-Commercial Discussion
>
> Message-ID: <519bcadf.1000...@cmsws.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 5/21/2013 11:54 AM, Ahmed Munir wrote:
> > Checked in /var/logs/ directory, all logs are not rotating by lo
<519b9fa6.9000...@lists.minotaur.cc>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 21/5/13 4:19 pm, Ahmed Munir wrote:> Last year, I installed Asterisk
> 10.4.2 and enabled logrotate on daily basis
> > which was working perfect. Now in couple of mont
Please advise so I can resolve this issue.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
on -1
Last updated: Fri Mar 29 12:25:10 2013
Stack: Heartbeat
Current DC: asterisk1 (887bae58-1eb6-47d1-b539-d12a2ed3d836) - partition
with quorum
Version: 1.0.12-unknown
2 Nodes configured, unknown expected votes
1 Resources configured.
Online: [ asterisk1 asterisk2 ]
uot;iso-8859-1"; Format="flowed"
>
> On Wed, 27 Feb 2013, Ahmed Munir wrote:
>
> > I'm getting compilation error as trying to install latest version of
> dahdi
>
> >
> /usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xdefs.
* [all] Error 2
Please advise how can I resolve this issue.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webi
gards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-
manually delete the original ARP entry before
> starting the ping).
>
> Pete
>
> --
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live int
> Can you be more specific about your Asterisk version? 10.xx.yy ?
>
> Sounds like some sort of resource leak.
>
>
> On Tue, Jan 15, 2013 at 3:02 PM, Ahmed Munir >wrote:
>
> > Hi,
> >
> > I configured Asterisk 10 for inbound fax, for couple of weeks I didn
the reason that I'm getting this message and how can
I avoid it?
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introdu
k/voicemail/default/'1234567/unavail.wav'));
Do I need to modify any other configuration? Please advise to resolve this
issue.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http
d on your system.
>
> DELIMITER @@ CREATE TRIGGER Test_Trigger AFTER INSERT ON MyTable FOR
> EACH ROW BEGIN DECLARE cmd CHAR(255); DECLARE result int(10); SET
> cmd=CONCAT('sudo /home/sarbac/hello_world ','Sarbajit'); SET result =
> sys_exec(cmd)
ltime.
Currently I've set a cron job that execute my script every 30 seconds and
checks for a new data in DB. If new data is inserted in 30 seconds that
script will run and sends the data to Asterisk for making calls. (This is
the case which I'm thinking to avoid)
Please advise.
--
allation-guide
>
> On Tue, Aug 14, 2012 at 7:20 PM, Ahmed Munir >wrote:
>
> > Hi,
> >
> > I would like to know, anyone who worked in Email to Fax scenario? If so
> > please share the idea for implementing it.
> >
> > As on other hand I configured Aster
Hi,
I'm looking for SIP client that supports T.38 Fax other than zoiper.
Please advise at earliest.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Ast
this case.
Please advice.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
advise to overcome this warning.
On 06/22/2012 12:05 PM, Ahmed Munir wrote:
> >
> > Here is my setup;
> >
> > Fax machine -> PSTN -> Cisco Voice GW -> IP cloud -> Asterisk. As on
> > Cisco Voice GW, T.38 fax already configured on SIP protocol.
>
> Appa
Here is my setup;
Fax machine -> PSTN -> Cisco Voice GW -> IP cloud -> Asterisk. As on Cisco
Voice GW, T.38 fax already configured on SIP protocol.
> Does your VoIP provider support t.38?
>
> Sent from my iPad
>
> On Jun 22, 2012, at 11:05 AM, Ahmed Munir wrote:
&
G T.30 ECM carrier not
found
As in sip.conf the configuration is listed below;
t38pt_udptl = yes,fec,maxdatagram=400
faxdetect = t38
And the rest are the standard configuration.
Please advise to resolve this issue.
--
Regards,
Ahmed Mu
Hi,
I would like to know whether SpanDSP supports T.38 for Asterisk 10? Because
as far as using Fax for Asterisk, I'm getting some issues using T.38
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provid
m for
T.30 faxing but I would like to know whether it also supports T.38
protocol or not?
Is there any other reliable method available for FoIP? If it is, please
share your views.
--
Regards,
Ahmed Munir Chohan
--
_
-- Ban
; fax machine to 9600(which is the lowest speed rate available fax machine),
> I was able to send 2 pages document fax but tried to send 3 pages document,
> I'm getting this error message.
>
> The Asterisk version I'm using is 10.4.2. Please advise me at earliest to
> overc
ax but tried to send 3 pages document,
I'm getting this error message.
The Asterisk version I'm using is 10.4.2. Please advise me at earliest to
overcome this issue
Note: Logs can also be provided as per request
--
Regards,
Ahmed Munir Chohan
--
___
On Tue, Jun 5, 2012 at 12:47 PM, Ahmed Munir wrote:
> I figured out the problem. Actually the sending fax machine speed was set
> as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully
> set to use T.38 protocol. After that I was able to receive fax.
>
> Thanks Ti
WARNING[10072]: res_fax.c:1687 receivefax_t38_init:
> > Audio FAX not allowed on channel 'SIP/192.168.1.69-0005' and
> > T.38 negotiation failed; aborting.
> > [Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error
> > initializing channel &#
assist me out to resolve this issue at
earliest.
>
>
> > Thanks for your response. Here is my topology as listing down below;
>
> > PSTN Line --> Cisco Voice GW --> IP Cloud --> Asterisk
>
&
't
want
to use Hylafax + iaxmodem as per requirement.
Please advice.
> Date: Fri, 1 Jun 2012 10:19:12 -0400
> From: Ahmed Munir
> Subject: [asterisk-users] Fax over IP ?
> To: asterisk-users@lists.digium.com
> Message-ID:
> >
> Content-Type: text/plain; cha
phone lines will the topology looks like as listed below;
PSTN Lines --> Asterisk (mounted a T1/ analog card) --> IP --> Asterisk
(receive Fax over IP)
or else?
Please advice.
--
Regards,
Ahmed Munir Chohan
--
_
--
s? because I'm getting very
irritated and need to set verbose level at least 4.
Further added, I also tried to stop and start asterisk service but still
getting these messages.
--
Regards,
Ahmed Munir Chohan
--
Regards,
Ahm
# in first
> character of line 86 and 102. Or modfy /etc/sudoers to allow your sudo to
> execute ulimit.
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
> Sent: Wednesday, February 29, 2012 8:5
rator assigned me sudo access for restarting asterisk
service.
Please assist me out to resolve this issue at earliest. I also tried to set
ulimit till 4-40 times higher than currently set i.e. 1024 but still giving
me same message.
--
Regards,
Ahmed Munir C
rator assigned me sudo access for restarting asterisk
service.
Please assist me out to resolve this issue at earliest. I also tried to set
ulimit till 4-40 times higher than currently set i.e. 1024 but still giving
me same message.
--
Regards,
Ahmed Munir C
lain; charset="iso-8859-1"
>
> On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir
> wrote:
>
> > Hi all,
> >
> > I'm getting one way audio when calling over the SIP trunk i.e. end device
> > B (remote end of SIP trunk) can hear device A (softphone regis
headers 0 lines) ---
Really destroying SIP dialog '
04ce1d566f1f17a221caba261e2af...@test.localhost.com' Method: INVITE
The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
--
Regards,
Ahmed Munir Chohan
--
_
sqlplus
>(oracle)
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Message-ID:
><69ecf8ff3230bc206837478422f97aad.squir...@webmail.i.frys.com>
> Content-Type: text/plain;charset=iso-8859-1
>
> Ahmed Munir wrote
t;
> Content-Type: text/plain; charset="us-ascii"
>
> What are the permissions on the module you are trying to run? (ls -l
> /var/lib/asterisk/agi-bin/module)
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digi
ascii"
>
> The module probably isn't readable/executeable from Asterisk
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
> Sent: Wednesday, January 04, 2012 10:45 AM
> To: asterisk-
user profile
and in my AGI), but when I tried to call my AGI (perl) in dial plan, it
don't get executed.
Please advise me to resolve this issue.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http
Is there other way around doing it instead of enabling debug and verbose
from logger.conf?
> Message: 1
> Date: Thu, 27 Oct 2011 10:36:08 -0400
> From: Ahmed Munir
> Subject: [asterisk-users] Check which client access Asterisk using AMI
> To: asterisk-users@lists.digium.c
request was made to
asterisk.
--
Regards,
Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
)
exten =>
h,n,AGI(printfaxresults.sh,${FAXSTATUS},${FAXREASON},${FAXREASONTEXT},${FAXRATE},${FAXRESOLUTION},${FAXFORMAT},${FAXCFFFORMAT},${FAXPAGES},${FAXID},${FAXEXTEN},${ElapsedFaxTime},FaxesSent.log)
Please advice, how can I send fax in image format using T.30
--
Regards
56 matches
Mail list logo
