Displaying 20 results from an estimated 800 matches similar to: "Question regarding digital card TE412p"
2009 Oct 21
4
Concurrent calls including mysql taking lot of time for execution
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
wait_timeout=60
query_cache_type=1
query_cache_limit=4M
query_cache_size=512M
interactive_timeout=120
2010 Jul 08
2
DTMF issues/redial tones with rfc2833
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
dialled. We also tried auto and info for dtmf but could not get the dtmf to
work reliably, can
2009 Oct 15
3
DS3 capacity calls using asterisk
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 any DS3
card in asterisk box so as to handle around 600 calls?
Thanks
Sandesh
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2010 Jun 11
2
asterisk log problem
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 call from the
log if there is any problem. Is there any way of splitting the full log into
parts
2007 Jul 31
1
Problems using TE412P and TDM400B in a IBM x3650
Another day, another apparant unexplained hardware incompatibility.
I have a TE412P and a TDM400B living quite happily in a whitebox using an
Intel motherboard:
http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm
I tried to move to an IBM x3650 system. It uses a slightly newer chipset,
but apparantly it's in the same family. The SE-7230 board has been EOL'd
and the
2010 Jun 25
2
Call drops on group paging asterisk - 1.4.22.1
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 become a serious
problem, can anyone through some light on this group paging senario?
Thank you very much
2010 Jul 21
1
Redial dtmf tones randomly...asterisk 1.4.21.2
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 switch and
all the phones are connected to this switch and there are no non sip devices
in the
2007 Apr 24
1
TE412P (T1/E1+DSP) digium card cause server crash
Hi all
I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them
configured as an E1 PRI connected to PSTN and another one configured as a T1
E&M connected to Avaya PBX. Each card only uses two ports, so there are 2 E1
lines and 2 T1 lines connecting to this server. The purpose of this server
is as a TDM trunk gateway that gets call from E1/T1 and then forward to an
IP-PBX
2010 Oct 20
1
Parked calls drop asterisk-1.4.22.1
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
randomly it tries to call SIP/5060 instead of SIP/2200 (where 2200 is the
extension number of the operator)
2008 Feb 08
0
Interoperability between TE412P and Eurotech PRI E1 GSM & CDMA Gateway
Hi,
I am about to purchase an Eurotech PRI E1 GSM & CDMA Gateway to operate
with my Asterisk's TE412P interface.
Anyone here has any experience of having this combination? Any success
or failure stories would be greatly appreciated.
Thanks in advance.
Ash
2009 Jul 20
0
No subject
device somewhere in your communication path, and since voice is picked up as
DTMF, some device is also set to listen for inband DTMF.
What is the origination source of incoming calls to your system?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote:
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip
2010 Mar 19
4
Call Drops while doing assisted transfer from remote location
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 as person B
hits transfer button again to transfer person A to C, the call is lost.
But in the
2008 Jul 02
3
Unable to switch input to xen from serial console
Hi all,
When i do ''xm dmesg'' the last statement says "*** Serial input ->
DOM0 (type ''CTRL-a'' three times to switch input to Xen)" (i have no clue
what''s that supposed to mean??) But when i press ctrl-a three times at
the serial console, nothing happens.
Iam using minicom to connect to the serial port of xen machine. Once
xen
2008 Aug 21
2
doubt on releasing domain pages
Hi,
I am trying to release domU pages from page_list and xenpage_list
after domU shutdown while retaining the rest of the domain information.
To achieve this in __domain_finalise_shutdown i call
domain_relinquish_resources. This is failing to release pages from
page_list for type PGT_l2_page_tables and crashing dom0.
To be specific, while testing on mini-os i saw that when
2010 Mar 12
0
Regarding - P-Asserted identity and Privacy - SOLVED
Hi All,
I got this figured out, when the privacy is ON at the other end of the
server and when we get the Invite message to the server connected to PRI's,
just take the details from the invite message in the Dial plan and send the
calls as anonymous:
exten => _1NXXXXXXXXX,n,Set(PRIVACY=${SIP_HEADER(Privacy)})
exten => _1NXXXXXXXXX,n,ExecIf($["${PRIVACY}" =
2008 Oct 01
3
GSM / 3g channel bank
More than 60% of our outbound calls are now to mobiles, so the time has
come to whack in a gsm channel bank.
Does anyone have any preference of bank ? Do you use a PRI or VOIP
connection from the bank to asterisk ? Real-world experiences are sooooo
much better than marketing blurb ;)
We currently have a TE412P with a free socket, so we have a choice
either way. I am looking for up to 30
2007 Feb 14
4
Guide to better performance using * ?
Can someone point me in the right direction to find documentation on
best practices when setting up a new Asterisk server? I'm using RHES4
and Dell 1750 with TE412P. My current problems are frequent crashes and
choppy audio so I think I can easily tweak these out of the picture.
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2009 Nov 30
3
Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.2.37,
1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases have been created in response to a SIP remote crash
vulnerability.
Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an
SDP regression
2009 Nov 30
3
Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.2.37,
1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases have been created in response to a SIP remote crash
vulnerability.
Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an
SDP regression
2009 Nov 06
4
problem while compiling asterisk tar file
hi friends,
i have installed asterisk,libpri,dahdi tar files in /usr/src.
problem is that when i compile (./configure) asterisk-1.4.26.3, gtk+2.0.0
dependency is missing. i installed gtk from gtk.org, now when i am
compiling gtk (./configure), i am getting this error message
configure:
WARNING: *** TIFF plug-in will not be built (TIFF library not found)
***
configure: error:
*** Checks for TIFF