Displaying 20 results from an estimated 1200 matches similar to: "Low cost hardware time for production environment"
2004 May 20
1
Grandstream tftp cfg.txt format
Hello!
I've been reading through the archives on this list for the last 8-10
months. There are some reports on success with tftp autoconfiguration
with a given cfg.txt format but really vague. Has anybody successfully
done this without using GAPS, or has anybody got a correctly formatted
cfg.txt file that works (from GAPS). I would be happy to write a script
or a java program that
2004 Jun 08
2
grandstream ringtones - makering.pl usage for 1.0.50
If you wan't to create a ringtone with makering.pl for firmware 1.0.50,
be sure to create it as ring.bin and then rename it to ring1.bin /
ring2.bin or ring3.bin. This seems to be the only change between the
format from 1.0.4.68.
Regards,
Maron
2005 Mar 24
3
Asterisk as Cisco Call-Manager - dial out to PSTN
Hi all,
I'm running Asterisk since two days, and it's really one of the phatest
software available on the net!!! Respect!!! I have connected Asterisk as a
call manager for a cisco gatekeeper. Everything works fine internal, but if
I want to ring to a PSTN over another call manager, which is connected over
ISDN, I get the following output. Has anyone experience in this or can help
me?
2003 Sep 09
1
Getting a local number abroad - Newbie question
Hello!
I have a staff member abroad and need to provide him with the ability to
make local calls. The features I need are:
* Possibillity to make calls at local (Icelandic) charges from Ireland
office.
* Possibillity to call the local Icelandic number and reach the Ireland
office.
I'm also wondering if there is any isdn based solution since there is a
possibillity of another staff
2005 Jan 12
2
Call Manager or Asterisk
Hello list.
No intention to start a flamewar here but I would really like opinions
from those who know both the Cisco and Asterisk system. I'm working for
a company with 15 offices in 11 countries, offices are relatively small
(3-20 people each) and most of them have a Cisco 1760 Router installed
with Call manager express (CME) and 1-3 ISDN lines (2-6 simultaneous calls).
We
2005 Jun 14
0
Snom hardware quality
Hi List.
I've used 3x Snom 200 in a hotel environment for a year, all of them
have failed at some point (1 went up in smoke, 1 handset failed, 1
display failed). I have 110 grandstream phones there of which only 3
have failed (whereof 2 due to brutal abuse), and in another company I
have 75 Cisco 7940 (SCCP Firmware) phones where not a single one has
failed (during 2 years).
I have no
2005 May 27
3
Recommended Network Latency
I'm planning on setting up some remote agents and before doing so, I
did some simple PING tests to measure latency. The average latency I
got was 250ms. Does anyone have experience in terms of quality of
calls when there is such high latency? Can anyone comment?
Thanks,
Waldo
2004 Jun 07
3
Voip-talk?
Hi everyone
I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments.
Cheers
Matt
2004 May 20
2
Fedora Core 2 and Kernel 2.6
Hi All,
I decided to have a go at installing Asterisk on FC2 which now runs on
Kernel 2.6..
Unfortunately I didn't get very far..
When trying to build zaptel it required me to link /usr/scr/linux-2.6 to
the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess
thats still the RH infulence.. :)
After than I tried again but the page rolls with errors and finally ends
with..
2011 Mar 10
1
Debian 6 / Xen 4 / blktap2
Hello all
I am updating a test system from Debian 5 and Xen 3.2 to Debian 6 and
Xen 4.
After dist-upgrading, I did:
aptitude -P install xen-hypervisor-4.0-i386 linux-image-xen-686
And then fixed Grub2 to boot the xen kernel.
cat /etc/debian_version return 6, uname -a returns:
Linux xen-test 2.6.32-5-xen-686 #1 SMP Wed Jan 12 07:52:18 UTC 2011 i686
GNU/Linux
So far so good.
I previously
2011 Dec 13
4
Keep sourcing when there is an error
Hello,
I want to know if there is any way to avoid source() stopping when there
is an error.
Here is the content of my Main.R script:
source("~/R/source/Constructor1.R") # Object1 should be constructed
ifelse(exists("Object1"), # It's an S4 object
print("Object1 exists"), # I can't avoid using 'validity'
2008 Oct 24
2
Asterisk and Cisco Call Manager Express (CME)
I was thinking about complicating my Voip setup by adding CME. I found
this example here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
and here: http://www.pasewaldt.com/cme/cme_index.htm
Would anyone like to comment on their experiences using CME with Asterisk...
I would like one of my Cisco phones to remain SIP connected directly
to my Asterisk system. The
2009 May 20
3
Asterisk CCM, CME Integration
Hi All,
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME
-----> 461X Phones
461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X
Phones
so in
2005 Jun 13
1
about timeouts
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi folks,
I've this infrastructure:
|voip services| -- |*| -- |cme| -- |isdn|
the voip services are logged on my *, then forwarded to number 601 on
cme. The isdn calls too are forwarded to 601. On cme I've a timeout X
for call-forward noan (no answer) to a specific number on * (5901) that
is my x-lite software client. If 5901 is
2006 Mar 01
1
Cisco Callmanager integration with asterisk
Hello
We have integrated cisco callmanager 4.1 with asterisk and we can dial from
cisco to asterisk but we're getting an error if we call from asterisk to
callmanager. This is the error I'm getting
anybody can help me?
Verbosity is at least 3
-- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack
-- Called cme-pbx/4455
-- SIP/cme-pbx-25ae is
2005 Mar 16
2
[Possible SPAM] : about sip, asterisk and cisco ccme
I am starting to work on a similar solution, but with full call manager
rather than CME. I am going to use Asterisk to accept POTS calls
through PCI FXO ports (winmodems) and then forward the calls through to
call manager via SIP. I don't have my FXO cards yet (waiting for UPS
man!!) but I have * talking to the CM through SIP just fine. I am
testing with the Cisco softphone, connected as a
2005 Aug 23
2
rsync problem
Hi,
My rsync is stopped working suddenly I got following in verbose and
log,
mkstemp failed: No such file or directory
and
rsync error: received SIGUSR1 or SIGINT (code 20) at rsync.c(229)
my rsync code :
rsync -az -e ssh --delete $HOSTTOBACKUP:$SOURCE
$DR_BACKUP_DIR/daily.0 >$tempfile 2>&1
the same code was working last week, what will be the problem, how
to proceed to fix?
2009 Jan 07
1
CISCO 7940 United_States/7960-tones.xml
I have a smartnet contract for this phone, and have searched high and
low for this file on the Cisco website.
I need:
United_States/7960-tones.xml
English_United_States/7960-font.xml
Every road seems to lead to the Call manager express downloads... I
don't have a CME, so that's basically useles.
Can anyone point me in the right direction?
Mikel
2009 Jun 05
1
DTMF Problem w/ MeetMe
First, the scenarios:
Call placed from Boston to locally configured Asterisk Meetme extension:
Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston)
Call placed from Boston to European Asterisk Meetme extension:
Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco
2821(CME,Europe) <-SIP-> Asterisk(Boston)
In the 1st scenario, everything works
2004 Aug 27
1
Help with a fax via Grandstream Handytone 286?
I have an analog Fax machine which I wish to connect to the network and
the Asterisk server. It will connect through a GS Handytone 286
converter and then into the LAN. Is there any information out there on
what I need to write in *sip.conf* and/or *extensions.conf* to make sure
the fax works as a fax?
Channel 8 on my T1 is a reserved, dedicated line for the fax number. Do
I need to