Chuck Bunn a écrit :
Hi,
I am planning on restarting asterisk nightly as I seem to be
experiencing some sort of memory leak (Asterisk slows down over time).
I have reviewed the Asterisk suggestions for management and one item
is the routine rebooting of Asterisk. Since I have Asterisk 1.2.1
Hi,
We are using Cisco5350 as a gateway with 2 E1 cards (part# AS535-DFC-2CE1)
to terminate calls. We need 2 more gateways, my question is can we save some
money and use Asterisk + PCI E1 cards?
If so, do you recommend any cards/configuration?
Thank you
Ahmed
Linuxnizer The Mesmorizer a écrit :
Hi,
We are using Cisco5350 as a gateway with 2 E1 cards (part#
AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my
question is can we save some money and use Asterisk + PCI E1 cards?
I've had the same issue lately. I need to set up a 4E1 /
BJ Weschke wrote:
On 12/16/05, Evert Meulie [EMAIL PROTECTED] wrote:
Hi all!
I am looking for a device that I can stick in a USB-port on my Asterisk
server and that allows me to connect one/more (cordless) PSTN-phones in such
a way that they'll work with SIP/Asterisk. I know
there are
We have a public folder full of contacts, but I understood that you
could only access this if the contacts were contacts in AD?
I was planning on doing a match on telephone number, mobile number and
fax. And then pulling a shortened version of the name as the caller ID,
Steve
-Original
Ignoring SS7, why exactly are you setting up several boxes ? there are quad
E1 cards no ?
This is way out of my league, but I just want to understand.
- Original Message -
From: Jean-Michel Hiver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Matt Florell ha scritto:
The best Dell for a production environment Asterisk server is no Dell
at all. They make some great workstations, but I've had many problems
with their servers(as have many others on this list) when trying to
use them in production for Asterisk. Take a look at the Digium
AR Tarzi a écrit :
Ignoring SS7, why exactly are you setting up several boxes ? there are
quad E1 cards no ?
This is way out of my league, but I just want to understand.
Because you would need a super monster box to do simultaneous g.729
encoding - and even though I'm not sure it would work
I need some information on the syntax used in features.conf.
I want to use the applicationmap to assign different buttons to the Hangup()
command. Where should I look?
Obelix
I want to use '##' to terminate a call instead of the '*' used by the Dial
command's H option.
Is there a way
From: Jean-Michel Hiver [EMAIL PROTECTED]
Linuxnizer The Mesmorizer a écrit :
Hi,
We are using Cisco5350 as a gateway with 2 E1 cards (part#
AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my question
is can we save some money and use Asterisk + PCI E1 cards?
I've had the
Hi
I need send a codenumber + key R (flash) from isdn telephone to a interface
on pstn.
isdn telephone -asterisk - (fxo)-- interface
Help me!!!
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
Hii Jean-Michel,
Couple of notes, I didn't find Audiocodes at voipsupply.com.
This is the product I'm going to order:
http://www.voipsupply.com/product_info.php?products_id=213osCsid=8afe5c480fd75d05ce6e5dad5876e3be
Final note, I can get a used Cisco5350 for around $7000 with 2E1
cards,
Hi!
Upgarde to 1.2.1 and try again - 1.2.0 (and maybe the beta) had a bug
concerning .call files and the non-passing on of variables that might
affect you as well.
Cheers, Philipp
Hmmm seems like every dialplan snippet I've seen so far relies on
ResponseTimeout and looping back to s,1. Is
i am using asterisk1.2.1 realtime mysql4.1.x
i found same update error in debug mode
i cat /var/log/asterisk/debug follow:Dec 13 00:12:28 DEBUG[7533] db.c: Unable to find key '99015' in family 'SIP/Registry'Dec 13 00:12:28 DEBUG[7533] res_config_mysql.c: MySQL RealTime: Update SQL: UPDATE
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use
Excuse me Chris!
Forgive me that I don't understand what you are really mean?
I would very appreciated if you let me know some think about the
rules,
and
how we would get help from people and how to find some previous
information
that has been posted from someelse before that we may
Is there any possible way to make TDM01B answers when
the other side pick up the phone ,and to prevent it
from answering just when it starts ringing?
Yes, if I understand your question properly.
Suggest you post relavent parts of zapata.conf and extensions.conf
that are associated with the
I suspect lots of people use the cid_rewrite script by Jay Milk. It's a
great script that updates the CID info by looking up callerid ID from
411.com (reverse lookup)
The script seems to be stuck at version 1 so I added a few enhancements to
bring it up to ver 1.1 The biggest is the addition of
50 extensions, 27 trunks, 1 queue, any tips would be great appreciated,
-Kerry
Inside op_style.cfg:
btn_width=191
btn_height=30
btn_padding=5
Then tweak all the scales and margin parameters for the icons. It
would give you all the buttons you need an a couple more.
You can direct all this
Hi all,
There is any possibility to have two local consoles using ALSA
devices?
I see no such an option in the alsa.conf nor extensions.conf files
Thank you and best regards,
Dan
___
--Bandwidth and Colocation provided by Easynews.com --
That TINYINT is probably the culprit then since the message is in
short. That code is converting the number to a 16bit short value. Are
there any other perl scripts that modify tables with TINYINTs in them?
From looking at the module, it doesn't look like it is reporting an
error, but just
Has anyone used a Cisco 7940/7960 (with or without a 7914) to display busy
extensions and if so, would you mind sharing the XML code to do it?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
--Bandwidth and
Hi,I am using a sjphone to connect via SIP to an Asterisk 1.2.1. For testing reasons I but the following in extensions.conf---cut---[from-sip]exten = 2000,1,Answer()exten = 2000,2,Wait(1)
exten = 2000,3,SayDigits(123)exten = 2000,4,Hangup()---cut---When ever I call the 2000 asterisk -vc
I am trying to get some feedback from anyone who may have experience of
a problem I am having.
We have several buildings that having only fiber to them so in order
that the alarm panel can call the central station, I have provided a
Sipura 1001 ATA. I can make a call to the central station
i can't register to my sip service.but x-lite can.
i think because my sip service domain is not really domain, they using sip proxy to resolve this domain
who can help me fix this problem thanks :)
look for follow line:
Asterisk SIP REGISTER header
Hi,
What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings. If you dmtfmode is set to 'inband' and you are using
anything other than ulaw or alaw codec it wont work. Also since your
hear the phone sometimes you may be experiencing QOS issues on your
network. Doe
I just had a setup like that; the alarm company is coming next week to
install and test. Make sure the panel is setup for DTMF and not for pulse;
I have found this is the case on some panels.
Gonzalo Gonzalez
- Original Message -
From: Chris Mason (Lists) [EMAIL PROTECTED]
To:
Hi Chuck,2005/12/17, Chuck Bunn [EMAIL PROTECTED]:
What are you codec and dmtfmode settings in sip.conf and in the sipphone settings.I use gsm.
If you dmtfmode is set to 'inband' and you are usinganything other than ulaw or alaw codec it wont work.I changed the settings and tried:---cut---exten =
I am trying to get some feedback from anyone who may have experience of
a problem I am having.
We have several buildings that having only fiber to them so in order
that the alarm panel can call the central station, I have provided a
Sipura 1001 ATA. I can make a call to the central station
Hi,
If you do not have QOS assigned to the SIP protocol it is quite possible
that there are packets time outs and the packets are discarded. Is it
possible to test the network during the evening or at a time when
traffic is at it lowest? Also try several traceroutes and see if there
is a
Hi,
Something else I should mention. Sip uses UDP and TCP packets. TCP
packets are used if there is congestion on the network. I am unclear
about what mechanism causes sip to switch between UDP and TCP but I
believe it is controllable - I believe It would be easier to use QOS
though. If UDP
I don't believe asterisk has any sip tcp support. Its all udp.
Hi,
Something else I should mention. Sip uses UDP and TCP packets. TCP
packets are used if there is congestion on the network. I am unclear
about what mechanism causes sip to switch between UDP and TCP
Hi,
Rich I stand corrected you are absolutely right - see
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
The following appears on the page:
Please note
* Asterisk does not yet support SIP over TCP. It only supports SIP
http://www.voip-info.org/wiki/view/SIP over UDP.
I'm wondering if anyone has ever implemented a scenario where calls aren't
terminated directly via Asterisk, but instead are passed back to a proxy, such
as SER to terminate the calls. With basic dialling, it would be easy. For basic
calling...
exten = XXX, 1, Dial(SIP/[EMAIL
This is a very big headache for me.
Alarms today normally use a protocol called contactID for
communications. These are very short dtmf tones and most devices have
a very hard time transmitting them. also any jitter etc causes them
to be unreadable. If anyone has a reliable method for
Exchange contacts != AD entries. Contacts in Exchange are basically email
messages with metadata. Now, if all of your contacts WERE in AD, you could
do a script to query AD through LDAP (that's what AD is - LDAP with MS
extensions) and you would solve latency problems when Asterisk would query
AD
On Tue, Dec 13, 2005 at 04:57:08PM -0500, Leah Newmark wrote:
Hi, All.
We recently installed Asterisk 1.2.1 through the Debian package/CVS.
Are those self-made packages or packages from Sid? What do you mean by
CVS?
If official packages, I suggest you reportbug(1) .
The CLI, however,
On Tue, Dec 13, 2005 at 05:21:42PM +, Karl O. Pinc wrote:
On 12/13/2005 07:32:10 AM, Kevin P. Fleming wrote:
This script is completely unnecessary on Debian; just add the modules
you wish to load into /etc/modules and they will be loaded at boot
time.
FYI the list. Using debian
On Wed, Dec 14, 2005 at 08:49:06AM -0700, Chuck Bunn wrote:
Hi,
I am planning on restarting asterisk nightly as I seem to be
experiencing some sort of memory leak (Asterisk slows down over time).
This is not an indication of a memory leak. The size of the asterisk
process:
ps `cat
On Thu, Dec 15, 2005 at 01:45:24PM +0100, Simone Cittadini wrote:
screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run
safe_asterisk in production
Any reason you need to run asterisk in a console?
asterisk -r allows you to view the current console.
/var/log/asterisk/messages
On Tue, Dec 13, 2005 at 06:26:49AM +, Karl O. Pinc wrote:
Hi,
Don't know if this is really right, all I know is
that Debian sarge does not have /var/lock/subsys/.
I foolishly made this patch against the zaptel 1.2
branch rather than trunk, although I did check that
the trunk has the
I'm sure this question has been asked before but I can't seem to find any
info on it.
Is there anything special that needs to be setup on the PAP2 side and the
Asterisk side for the PAP2 to work on the asterisk server.
I've entered all the settings for my VoIP provider but all I get is
On Tue, Dec 13, 2005 at 03:24:33PM -0500, Vladimir Montealegre wrote:
i have two cards md3200 buy they dont work is possible connect two single
phone lines with 2 cards x100 clone ??
Basically, just as you can connect two phones on the same line: not
together. In practice: think of a line with
Tzafrir Cohen wrote:
BTW: some modules may provide a span (/proc/zaptel/n , not necessarly
/proc/zaptel/1) but not function as a timing device. In which case
you'll still need to modprobe ztdummy, right?
That would be true, although none of the drivers in the Zaptel source
distribution fall
On Mon, Dec 12, 2005 at 11:28:35AM -0800, Johnny Voice wrote:
For my asterisk installation in my lab, I will install the
RedHat
Linux ES v4
distribution (with kernel 2.6) onto a Dell Power Edge 1650 with
~16GB of Raid-1 hard disk space.
Not much. Asterisk on its own doesn't take much
Has anyone been
successful getting Auto-Answer by Call-Info to work with the GXP
2000
I have followed the
suggestions in
http://www.voip-info.org/wiki/view/GXP-2000
Specifically I
have:
1. Upgraded to
1.0.1.13, which supposedly supports this feature
2. Set Allow
Auto-Answer by
Running Asterisk 1.2.1 on Suse 10.0 X86-64.
Tried to get mpg123 0.59r which came with the 1.2.1 dist running on
this box, but all I get is poop:
as -o decode_i586.o decode_i586.s
decode_i586.s: Assembler messages:
decode_i586.s:44: Error: suffix or operands invalid for `push'
/home
An asterisk system typically does not have users and need nt have a
separate /home
I disagree here.
You have at least 1 user to remotaly login to the system to
do some work on it. Think config changes etc.
In case of unauthorized access (ppl stole your password or
whatever) you will
Hi:
i have these configured in zapata.conf:
signalling=fxs_ks
context=incoming
channel = 1
and these in extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DeadAGI(astcc.agi)
exten = s,3,Hangup
[tele]
exten = _01XX,1,Dial,ZAP/1/${EXTEN}
for example when i try to dial [EMAIL
Hi,
Thanks for the input. I will try your suggestions. By slowing down the
server takes longer and longer to respond to prompts such as retrieving
voice mail. I am recompiling my install this weekend as I have had a
continued problem with logs (see other post) and this might be related
to
Hello,
I read
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
So i set callgroup and pickupgroup in sip.conf .
How can I forward an incoming call to one or more
callgroup.
Regards
Harry
On Saturday 17 December 2005 15:18, Michiel van Baak wrote:
I disagree here.
You have at least 1 user to remotaly login to the system to
do some work on it. Think config changes etc.
In case of unauthorized access (ppl stole your password or
whatever) you will be glad you have /home on a
On Saturday 17 December 2005 15:23, chawki hammoud wrote:
[tele]
exten = _01XX,1,Dial,ZAP/1/${EXTEN}
for example when i try to dial [EMAIL PROTECTED] the call
is been answered when it starts ringing and not when
No, the call is *not* answered when you hit this line in the dialplan. If
hi,
Do anyone have experience with the Sangoma E1 A102 or A104 etc?
I am tempted to buy one for testing out, but I don't want to waste more
money and find that they have the same issues as the Digium's.
I know Sangoma have a better solution to IRQ problems, but I know
nothing about their
On Saturday 17 December 2005 16:21, [EMAIL PROTECTED] wrote:
I am tempted to buy one for testing out, but I don't want to waste more
money and find that they have the same issues as the Digium's.
They work about the same. I've never had IRQ issues with Digium though (even
sharing IRQs).
I
No I do not believe so. Zaptel's pretty strict about keeping the amount of
queued data to an absolute minimum.
Do you know if this is a driver or hardware limitation?
Jan
___
--Bandwidth and Colocation provided by Easynews.com --
HI:
I dial this on console :
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp, zap/1/01472345) in
new stack
-- Called 1/01472345
-- Zap/1-1 answered OSS/dsp
Console call has been answered
the call here to 01472345 is been answered before the
other side (01472345 side) pick up the
On Sat, Dec 17, 2005 at 09:18:39PM +0100, Michiel van Baak wrote:
/home
An asterisk system typically does not have users and need nt have a
separate /home
I disagree here.
You have at least 1 user to remotaly login to the system to
do some work on it. Think config changes etc.
In
Hi Jason,
I've got several PAP2s working with asterisk. Feel free to e-mail me
off-line if you want to compare configurations. Which version of
asterisk and which PAP2 firmware are you running?
Cheers,
john
Jason (WeatherServer) wrote:
I'm sure this question has been asked before but I
Hello,
I added callgroup=1 and pickupgroup=1 for sip channels
however I can't pickup a call (see below ) between sip
phones when i dial *8 .
May I have to add app_pickup to solve this problem.
I use asterisk-1.2
Regards
Harry
serveur1*CLI
-- SIP read from 80.119.8.167:5060:
ACK sip:[EMAIL
On 00:03, Sun 18 Dec 05, Tzafrir Cohen wrote:
On Sat, Dec 17, 2005 at 09:18:39PM +0100, Michiel van Baak wrote:
/home
An asterisk system typically does not have users and need nt have a
separate /home
I disagree here.
You have at least 1 user to remotaly login to the system
On 15:41, Sat 17 Dec 05, Andrew Kohlsmith wrote:
On Saturday 17 December 2005 15:18, Michiel van Baak wrote:
I disagree here.
You have at least 1 user to remotaly login to the system to
do some work on it. Think config changes etc.
In case of unauthorized access (ppl stole your password
Teliax users,
I have a couple questions about Teliax, just
hopeing some current customers might shed some light on them.
How reliable is a toll-free number from Teliax? Has
anyone had any problems with it?
The Pay as you go plan has a Billing of 60/1, what
does that mean? My guess is 60
Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom
From that article:
There is an 'allpage.agi' now available at
http://aussievoip.com.au/allpage.agi. Documentation is available in
the file. This should work with Snom and Grandstream GXP2000 phones
(and possibly budgettones if they
Ryan Burke [EMAIL PROTECTED] writes:
Is there any other charges because of the toll free number?
I was toying with the idea of getting an 800 number too, but the issue
of a substantial per call fee for pay-phones calls has me worried.
Hopefully someone here can clarify what the deal is there.
Looking for a good toll free DID provider. Any suggestions?
All ready tried Sellvoip and Gafachi and the experience was not desirable.
Thanks,
Tom Vile
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
You might have to use *8#. At least I do with my 7960.
I added callgroup=1 and pickupgroup=1 for sip channels
however I can't pickup a call (see below ) between sip
phones when i dial *8 .
May I have to add app_pickup to solve this problem.
I use asterisk-1.2
wolfgang,
Thanks for the heads up. I'm hoping to get some feedback from Teliax
toll-free customers and see if they would recommend the service. Plus I have
those few questions on billing.
Thanks again,
Ryan
- Original Message -
From: Wolfgang S. Rupprecht [EMAIL PROTECTED]
To:
I known that sip channel should be free from echo. I am find this is not the case for me.
The setup here is
Sipura 3000 connected to vonage
extensions are SIPURA 841 or SIPURA 2002 ATA.
I am getting echos on some of the outbound calls. I would like to be able to have one
of the software echo
I have a couple questions about Teliax, just hopeing some current customers
might
shed some light on them.
How reliable is a toll-free number from Teliax? Has anyone had any problems
with it?
They have been very reliable for me. Once in a great while they'll have
a problem, but then
Is there any other charges because of the toll free number?
I was toying with the idea of getting an 800 number too, but the issue
of a substantial per call fee for pay-phones calls has me worried.
Hopefully someone here can clarify what the deal is there. I've seen
numbers quoted as
No I do not believe so. Zaptel's pretty strict about keeping the amount of
queued data to an absolute minimum.
Do you know if this is a driver or hardware limitation?
Its not a limitation. Its an architectural design which is based on pulse
code modulation (pcm) standards, which
I have gotten the tftp server working and the 9133i is doing a firmware
update and finds the aastra.cfg file as well as the 00XXX.mac file. The
issue is that I can't figure out what is wrong in the configuration files
that it is not loading the extension, proxy, etc. info.
Could someone post
Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom
From that article:
There is an 'allpage.agi' now available at
http://aussievoip.com.au/allpage.agi. Documentation is available in
I'm the author of that, and I've actually re-written it, because I was
pretty unhappy with the
*sigh* Analog Zap FXO ports consider the call answered as soon as
it's finished throwing the DTMF at the telco. This is because a Zap
port CAN'T tell when an analog call has been answered.
Andrew Kohlsmith wrote:
On Saturday 17 December 2005 15:23, chawki hammoud wrote:
[tele]
exten =
On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote:
*sigh* Analog Zap FXO ports consider the call answered as soon as
it's finished throwing the DTMF at the telco. This is because a Zap
port CAN'T tell when an analog call has been answered.
Bah, you're absolutely correct. I
Rich,
Thanks for your feedback. Sounds like what I was looking for. I think I'll
sign up tonight!
Thanks,
Ryan
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday,
Andrew Kohlsmith wrote:
On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote:
*sigh* Analog Zap FXO ports consider the call answered as soon as
it's finished throwing the DTMF at the telco. This is because a Zap
port CAN'T tell when an analog call has been answered.
On 12/16/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
C F wrote:
Kevin, I'm not sure this would work here, but maybe it would.
There was a bug posted about being able to use hint against local
channels, would that not help him?
http://bugs.digium.com/view.php?id=5779nbn=4
No, the
On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote:
$1k for a single port T1
I've gone down the Tellabs route, and am infinitely more happy.thanks C F
for the docs..
-D
NP, anytime :) You seen the pics of the 253C? it's on the wiki. I'm
still looking for detailed docs on that.
On 12/16/05, Steve Davies [EMAIL PROTECTED] wrote:
On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote:
$1k for a single port T1
I've gone down the Tellabs route, and am infinitely more happy.thanks C
F for the docs..
Tellabs looks a little too up-scale for what I need :). $1k
Anyone have an indications.conf entry for Japan?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
On 12/16/05, Rich Adamson [EMAIL PROTECTED] wrote:
OK we need some help in setting up a good wiki-info page for setting up the
Mediatrix
1204 to work with asterisk. If anyone has
set these unit's up and have them working please post your settings here so
we can
create a page on the
Hey, I´m trying to modprobe ztdummy, but when i make modprobe, return one error.
I use kernel 2.4 and have UHCI USB Controller allowed in my kernel.
This problem can be, because i dont have any pci card (fxo) at the computer ?
Thanks.
___
--Bandwidth
Before I start hacking this into asterisk 1.2.1 I would like to known
if others are running into this kind of problem ?
Asterisk doesn't do any echo cancellation in the setup you describe;
it just passes the audio data, and transcodes if necessary. The
endpoints (the 841 phone and the 2002 and
modprobe, return one error.
What is the error?
Ztdummy is an alternative if you don't have a hardware timing source,
so not having a PCI FXO card is not the cause.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
86 matches
Mail list logo
