Hi
We are facing a problem for orphaned parked calls, we have the following
config:
asterisk -1.4.22.1
dahdi-linux-complete-2.2.0.2+2.2.0
and when we get an incoming call and after it gets parked, after some set
time (here its 2 min), it goes back to the operator, but the problem is that
Hi..
We are facing a problem that is making the channel to be stuck. we are using
asterisk 1.4.22.1 version, and we have a direct sip trunk...We have 2 queues
and one has 2 agents and the other 5 agents, from last week the second
queue's channel is getting stuck, it happened 3 times till now and
Hi,
We are having the redial dtmf tones issue generated randomly in the Voip/SIP
calls, [versions: asterisk 1.4.21.2, dahdi 2.0.2.2] and we have dtmf as
inband in the trunks. We actually have mutiple locations one server at our
datacenter, and from those locations people are complaining that they
, Benny Amorsen
benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
wrote:
I would appreciate it if you didn't top-post.
das sandesh sandesh...@gmail.com writes:
Hi Benny...
DTMF tones are heard at the SIP phones side and not the other
party...'server side' means from the Asterisk side
Hi,
We are experiencing this issue of redial dtmf tones generated randomly in
the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as
rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one
FXS is used for Fax and rest are empy) connected to the netgear
) inbetween the conversation...
Thank you
Sandesh
On Wed, Jul 21, 2010 at 2:31 PM, Benny Amorsen
benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
wrote:
das sandesh sandesh...@gmail.com writes:
In the wireshark capture attached we could see the random dtmf
digits have been sent from
share
their opinion on this...Thank you.
Regards
Sandesh
On Thu, Jul 8, 2010 at 5:21 PM, das sandesh sandesh...@gmail.com wrote:
Thanks Zeeshan.that server is located at the headquaters and phones are
at different locations, even with default rfc2833 mode, other party IVR
prompts
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip trunking for them
not PRI's, one of our system is giving a problem of dtmf (rfc2833), like
when we dial the number that have IVR and enter the extension or access
code, it some time takes it and some times does'nt recognize the digits
--
www.ilovetovoip.com
On 2010-07-08 4:24 PM, das sandesh sandesh...@gmail.com wrote:
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip trunking for them
not PRI's, one of our system is giving a problem of dtmf (rfc2833), like
when we dial the number that have IVR and enter the extension
, 2010 at 12:02 PM, Mike l...@net-wall.com wrote:
The phone brand and model might matter here, I have had no such problems
with Polycom phones.
Mike
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *das sandesh
*Sent
Hi All,
We are using group paging and our asterisk version: 1.4.22.1, but when ever
any one page to the whole group (28 extensions), the calls which are on hold
on those extensions will be dropped, is there any other way to have this
feature or to go with Overhead paging. Currently this has
-mtime +7 -exec rm
-rf {} \;
/usr/sbin/asterisk -rx logger rotate
exit 0
--
Dean Hoover
On 6/11/2010 11:08 AM, das sandesh wrote:
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where there would
be around 322 concurrent calls going on, but I can see that full log
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where there would be
around 322 concurrent calls going on, but I can see that full log grows
rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug
and its tedious even by using any commands to get the required
Hi all,
We have our system hosted publicly and 4 phones are connected remotely at
employee's home, and when they try to do a assisted transfer to one of the
employee at the main office, the call is lost. For ex: person A calls person
B, person B calls person C for assisted transfer, and as soon
=${SIP_HEADER(Privacy)})
exten = _1NX,n,ExecIf($[${PRIVACY} = id]|SetCallerPres|prohib)
This makes the calls with privay ON sent as anonymous at the other end. One
more thing is to make sure you enable usecallingpres=yes in chan_dahdi.conf.
Thank you
Sandesh
On Fri, Mar 5, 2010 at 11:18 AM, das
Hi All,
We have two servers, one server (SIP asterisk server) sending calls to the
second server(has PRI) which goes our through the PRI's (using TE 412p).
When the pprivacy is enabled: P-Asserted-Identity Header, privacy id are
sent in the header of SIP invite packet to the second server, how
and found that
version 1.4.17/18.1 had the issue of channel stuck up as well as random
asterisk crashes.
Regards
Sandesh
On Thu, Feb 11, 2010 at 6:29 AM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
das sandesh wrote:
Hi,
Asterisk got stopped this morning after 20 minutes and phones went
Hi,
Asterisk got stopped this morning after 20 minutes and phones went to 'No
Service' and then got started automatically after 20 min, as I could see in
the full log that asterisk got started at so and so time:
[Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started
Thanks for all your inputs...
Jeff:
I used the converter to convert the sound file to 8KHz, 16 bit PCM encoded
mono wave file which asterisk needs and now the POTS call quality is lot
clear than before but the cell phone is still the same, not much
clear...i think because of its voice codec
Hi All,
I tried using some music on hold (music) files, when I test it with normal
SIP phone its clear and good, but when I call from my cell phone or POTS
line it sounds a bit scratchy/static and not clear at all, is there any
software that i need to install in the asterisk system to make this
Hi,
I was trying to use 2 of asterisk servers and interconnected, one of them as
a peer to other sever (configured in sip.conf), so all the calls to server 1
will just be passed to server 2 (has PRI Card, TE 412P, only one PRI
connected), i was sending calls to server 1 and that would send to
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am planning to use TE412p (includes echo cancellation) 4 port digital card
(PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
connections) with proper hardware like dual core quadcore processor
TE410P with heavy load in a Pentium4 3,2GHz system running
Asterisk 1.2 was no problem. It did only IVR and bridging with no
transcoding though.
Chris
2009/12/14 das sandesh sandesh...@gmail.com:
Hi,
I was able to implement T122p one port PRI and was able to call out, but
I
am planning
Hi All,
I am having a problem with the ring group where when an incoming call comes
it rings all the 3 extensions associated to that, but intermittently it
rings one extension only once, but the others will be continuously ringing
and the goes to generalized voicemail. When I check the log using
Hi Alex,
I am using Ring All channel strategy...
Thanks
Sandesh
On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov
abalas...@evaristesys.comwrote:
What is the channel technology in use?
das sandesh wrote:
Hi All,
I am having a problem with the ring group where when an incoming call
We are using SIP channel technology...
On Tue, Nov 24, 2009 at 11:03 AM, Alex Balashov
abalas...@evaristesys.comwrote:
I am talking about the endpoints (extensions). SIP? DAHDI? IAX?
H.323?
das sandesh wrote:
Hi Alex,
I am using Ring All channel strategy...
Thanks
Sandesh
Thanks Dave, I added a new SSD harddrive instead of a normal SATA harddrive
as well as included my ip in the hosts file also I have included
'skip-name-resolv' as you mentioned to not to resolv and tested for around
250 concurrent calls, connection was going through fine...next week I
would be
Hi All,
I was actually trying to use the dialplan application that uses 'Dial' and
when the: Dial(SIP/xxx...@|20|) command is executed and the
destination number rings for 20 sec after which I receive as 503 Service
Unavailable, but not 480 Temporarily unavailable.
On Thu, Oct 22, 2009 at 6:20 PM, Benaiad bena...@gmail.com wrote:
Abdulmnem Benaiad
Almontaha CTO
Almontaha IT Co.
cell: +218 92 5200025
fax: +218 21 4835263
www.almontaha.com.ly
On Wed, Oct 21, 2009 at 11:57 PM, das sandesh sandesh...@gmail.comwrote:
Hi Matt,
I already used
Thanks for the information, I will look into both cisco and adtran see which
would be helpful
On Thu, Oct 15, 2009 at 4:09 PM, Alex Balashov abalas...@evaristesys.comwrote:
David Backeberg wrote:
On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com
wrote:
There's no
Hi,
I tried getting our server setup for 400-500 simultaneous calls, calls were
going through properly but at around 200-250 calls, mysql (connect ...)
statement was taking at least 5-10 sec to connect to the database. I
optimized all possible parameters in my.cnf:
max_connection=1000
be is it good to
go with dual quad core processor instead of just one inorder to handle the
call capacity as well as connections?
Regards
Sandesh.
On Wed, Oct 21, 2009 at 2:21 PM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 21 Oct 2009, das sandesh wrote:
I tried getting our server setup
PM, Matt Riddell li...@venturevoip.com wrote:
On 22/10/09 7:30 AM, das sandesh wrote:
Hi,
I tried getting our server setup for 400-500 simultaneous calls, calls
were going through properly but at around 200-250 calls, mysql (connect
...) statement was taking at least 5-10 sec to connect
There were 2 problems that we faced, one was at around 50 calls, few calls
were just dead air, and when I saw the logs I could see that it was sent to
the sip provider and after that there was no log for that particular call
that was having dead air, but at around 200 to 250, we could see that
You have to check and verify the SIP trunk details, as ext to ext works once
the pbx is up, but to call out, it should go through your provider.so
just recheck your provider's details.
Regards
Sandesh
On Wed, Oct 14, 2009 at 8:24 AM, Ott Rose sixfourimp...@hotmail.com wrote:
here is the
Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from digium but
unable to get this product. Does anybody have any idea of having
to limit the
concurrent calls.
Thanks
Sandesh
On Sun, Oct 4, 2009 at 9:06 PM, Matt Riddell li...@venturevoip.com wrote:
On 3/10/09 3:55 AM, das sandesh wrote:
I am using the command:
./sipp -sn uac -d 200 -s repective context pattern IP Address -l
200
Its 10 calls per second and 200
, Matt Riddell li...@venturevoip.com wrote:
On 2/10/09 12:41 AM, das sandesh wrote:
Hi Matt,
When I get can more that 150 calls, i get a busy signal (Congestion) for
the calls above 150 - says your call cannot be completed now, its
allowing only 150 callsIs there any thing related
:23 AM, Matt Riddell li...@venturevoip.com wrote:
On 3/10/09 2:40 AM, das sandesh wrote:
These calls are from asterisk. I am using sipp to generate the calls and
I increased the limits using the command line interface and used ulimit
-n 1 and as well as changed in /etc/security
the call
capacity.
Thanks
Sandesh
On Thu, Oct 1, 2009 at 4:19 AM, Matt Riddell li...@venturevoip.com wrote:
On 1/10/09 5:56 PM, das sandesh wrote:
Hi All,
I have a problem, when I was doing a performance testing using an
asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150
Hi All,
I have a problem, when I was doing a performance testing using an asterisk
server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls all the
other calls are giving busy, I tried to do ulimit related stuff, like
increasing the soft and hard limits to 10 but no luck, Any ideas
Thanks Darrick, I will try to upgrade the version. Also I got to know that
if we are going to limit the call length as well as silence detection might
me useful for eradicating the channel lockup.
On Fri, Sep 18, 2009 at 11:52 PM, Darrick Hartman dhart...@djhsolutions.com
wrote:
das sandesh
Hi All,
Today I faced a problem with channels getting stuck. We use asterisk
1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try
to do soft hangup channel, it says Requested for soft hangup for that
channel, but if we go and check once again those channels are still stuck.
Hi All,
I had a problem yesterday, that our asterisk server showed 2 channels were
in use continuously, but nobody were using any of them at that time. I had
to kill them using softhangup and I checked all the logs but could not
find why exactly this problem occurred, as the system was running
timeout is 1200, call will be disconnected after 20 min, so
what would be the better option for this cause of SIP channels stuck up
Thanks
Sandesh
On Thu, Aug 13, 2009 at 3:32 AM, Steve Howes st...@geekinter.net wrote:
On 13 Aug 2009, at 07:51, das sandesh wrote:
Hi All,
I had
->
[asterisk-users] multiple contexts for multiple locations
asterisk-users
-- Thread --
-- Date --
[asterisk-users] multiple contexts for multiple locations
das sandesh
Re: [asterisk-us
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