On Fri, Mar 03, 2006 at 11:39:49AM -0600, Kevin P. Fleming wrote:
> Matt Schulte wrote:
> > All, I'm not sure how to word this question but we're noticing a lot of
> > our asterisk boxes no longer have multiple asterisk child processes.
> > i.e. doing a 'ps ax' reveals only 1 asterisk PID when norm
Jordan-
I'm not sure if you found the files and instructions
on www.fitawi.com/Asterisk/. If you did I can offer you a
full refund of the purchase price. (oh right, it's free, I forgot)
I'm afraid I did make some assumptions about which packages
are installed on the system. If
Matt Schulte wrote:
> All, I'm not sure how to word this question but we're noticing a lot of
> our asterisk boxes no longer have multiple asterisk child processes.
> i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used
> to seeing 8+ .. There is no rhyme or reason to it, and we'
Aaron Daniel wrote:
> Thanks :) When we were using Mark2 with aggressive suppression, we had
> zero problems, but decided to go with the hardware canceler in our new
> gateway since hardware's supposed to be better than software...
> hopefully this works for us too.
'aggressive suppression' is hal
On Fri, 2006-03-03 at 10:45 +0100, serge messa wrote:
> Hi all
>
>I'm a newbie in asterisk.I install asterisk server
> successfully. I configure this server to traverse NAT.
> Using Xlite clients, i make a call between 2 local
> networks through Internet.Asterisk server is
> installed on a ho
Its good for us to post thing about different companies customer
service. I feel one of the most important points when buying a product
is if the company is going to stand behind it. I had purchased some
sipuras that did not work but was lucky and able to send them back to
the store I ordered t
only for the whole cardthe tx and rx gain affect all 24 channels.
-D
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Fri 3/3/2006 11:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo Cancelation on
Whisker, Peter wrote:
Can someone please tell me if it's possible to select the G726 codec
bandwidth for an IAX trunk between two Asterisk 1.2.5 servers?
I can select "disallow=all / allow=g726" but I think it defaults to the
g726-32 variant.
Is there any way of forcing Asterisk to use g726-
I had the same issue with issue with Sipura. I went though email
support. They finaly said email another address to get a RMA. That
support made me go though everything the 1st one did again. And in the
end they never responded after they decided I needed a RMA.
So it never got resolved.
In Asterisk the Agent / Queue setup is kinda different than most people may
expect. You can use a Queue without using Agents and Agents can be used without
Queues. Agents however extend normal channels with the ability to
login/logout/pause that is not available on Zap/SIP/IAX/etc.
I assume
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote:
The best way to achieve maximum manageability is to design a MySQL database and
develop AGI scripts (in your language of choice) that work to that design. I've
found that it has been far easier to develop complex routing logic
Can someone please tell me if it's possible to select the G726 codec
bandwidth for an IAX trunk between two Asterisk 1.2.5 servers?
I can select "disallow=all / allow=g726" but I think it defaults to the
g726-32 variant.
Is there any way of forcing Asterisk to use g726-24 for such a trunk
ex
Dumpolid Exeplish wrote:
> i am taking overr the administration of an existing production * PBX
> but i cant seem to find out which version of * this is. When i use the
> 'show version' coomandat the cli, i get this:
>
> Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
> 200
Is anyone using asterisk with the Zapata hardware and a Sprint FNTM(I
have seen this spelled fantom) T1 line?
I can take calls, but:
1. I don't know the correct way to get ANI/DNIS (*ANI*DNIS*)
2. I cannot place outbound calls.
From a custom app on another hardware/software platform I know from
ex
On a 55 station
install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are
complaiining about echo. According to the users, the echo seems to be phone
number dependant. They claim that certain phone numbers have echo while others
dont. Are there any tuning parametes like there
I've got a problem with chan-misdn
I'm using asterisk with a hfcsusb-adapter in nt-mode connected to an
isdn-telephone
making calls to other internal clients like sip or sccp are without
problems
if I call into (or receive a call from) the pstn via a zap-channel (Digium
E1-card) my outgoing aud
i am taking overr the administration of an existing production * PBX but i cant seem to find out which version of * this is. When i use the 'show version' coomandat the cli, i get this:Asterisk CVS-HEAD built by [EMAIL PROTECTED]
on a i686 running Linux on 2005-08-10 05:57:59 UTCcan anyone tell me
Anyone have any luck RMAing a Sipura phone since the Cisco take over?
Sipura only has support via email or fax to end users and I haven't
gotten a response to either for over 2 months.
Linksys Support will jump you through all their scripted hoops to
resolve your problem (they hope if they speak w
I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811..
On 3/2/06, Johnathan Corgan <[EMAIL PROTECTED]> wrote:
Gary Richardson wrote:> Now it seems that if I'm really l
Cisco has a book about it: http://www.amazon.com/gp/product/1587050609/sr=8-1/qid=1141401215/ref=pd_bbs_1/103-7257053-6939020?%5Fencoding=UTF8
While this isn't specifically about the SIP image, the XML browser is the same. I also use Cisco::IPPhone (http://search.cpan.org/~mrpalmer/Cisco-IPPhone-0.
I will
try to test your adaptation.
How I
congfigure to enable VAD?
Regards
Jsalas
-Mensaje original-De: Moises Silva
[mailto:[EMAIL PROTECTED]Enviado el: Friday, February 17,
2006 11:26 AMPara: Asterisk Users Mailing List - Non-Commercial
DiscussionAsunto: Re: [Asteris
A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez <
[EMAIL PROTECTED]> wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?
Amaur
Hallo!
I have problem with incoming calls on 2 phone
numbers registered on same SIP provider account. I've tried averything and
nothing seems to work. No matter what I do asterisk system refuses differ betwen
them and both got connected to the same extensions.
I've tride with:
registrat
A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez <
[EMAIL PROTECTED]> wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?
Amaur
Does anyone have a good resource to learn how to program the
soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware…thanks.
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Hello there,
I'm successfully using Asterisk Realtime to access information about
voicemail users from a MySQL database. Now I'd like to read static
voicemail information (such as format, serveremail, etc.) also from a
database. Is that possible? If so, I'm assuming one would need to attach
Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?Amaury Rodríguez http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 Go OpenSource And Be FREE!!
Yahoo! Mail
Bring photos to l
The instructions on the wiki for asterisk Realtime give the extensions
schema with the priority field set to be tinyint(4). This of course
cannot hold the value 'hint'
The question I have, is the solution simply to set the field to
varchar(n) as that will then hold 'hint' or any integer value
I have not (yet) had one of my bare bones systems lockup.. but they
also don't do 400-700 calls a day.
The system that does lockup experiences exactly the issues described
here which are you can connect to it asterisk -r and issue commands
but nothing responds not even stop now... and you have to
Hello all!
On Thu, Mar 02, 2006 at 03:12:05PM -0600, Joseph Tanner wrote:
> The problem isn't that asterisk isn't running, it's that asterisk is
> not responding. When asterisk is in this funky state, I can still run
> "asterisk -r" from the command line and get access to the CLI.
> While in
Sorry. Miss type 'can'. I meant 'cannot'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Martin
Joseph
Sent: Friday, March 03, 2006 12:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: G729 and Meetme
On Ma
Hello,
For a whole lot of different reasons, I am thinking of moving from pure VoIP
(my DID provider gives me SIP access and my termination is SIP too) to PRI
(possibly keeping termination in VoIP for long distance). FYI, my business
is Hosted PBX...and my end-points will stay SIP.
Here is the t
When you type 'help' from the CLI, it says nothing about 'quit' - or at
least not between 'no debug channel' and 'realtime load'. Google told
me about it, and I probably should have guessed, but still...
Who do I report this to?
Bob McDowell
___
-
Sean Cook wrote:
First things first... use the latest version... (that I know of)
http://www.fitawi.com/Asterisk/
second... which part are you having problems with? The web piece? or
the app_cbmysql?
For the app_cbmysql, I have found that the easiest way to work with it
is to incorperate it
[EMAIL PROTECTED] wrote:
Hi All,
I want to configure fax with Asterisk and I found that we can do this reliably
using G711 codec only. Currently my provider is supporting G729 and G711.
During the call initiation the call starts with G729 (1'st priority) and
Faxing via VoIP is not reliable per
Hi,
Is there a command (to use in a dial plan), to check the
call status during a call.
Kind Regards,
Arjan Kroon
Mobillion B.V.
Copernicuslaan 30
Postbus 554 / PO Box
554
6710 BN Ede
tel: +31 (0)318-648920
fax: +31 (0)318-648839
mobile: +31 (0)6-55871460
email: [EMAIL
I haven't tried sip yet... been finishing voicemail, but the principal
is the same.
res_mysql.conf
[general]
dbhost = localhost
dbname = asterisk
dbuser = someuser
dbpass = somepass
dbport = 3306
dbsock = /var/run/mysqld/mysqld.sock
extconfig.conf
voicemail => mysql,asterisk,voicemail
; i would
Hi All,
I want to configure fax with Asterisk and I found that we can do this reliably
using G711 codec only. Currently my provider is supporting G729 and G711.
During the call initiation the call starts with G729 (1'st priority) and
somehow if the receiver is unable to receive call then we are pr
First things first... use the latest version... (that I know of)
http://www.fitawi.com/Asterisk/
second... which part are you having problems with? The web piece? or
the app_cbmysql?
For the app_cbmysql, I have found that the easiest way to work with it
is to incorperate it into asterisk-addon
This has to be the worst documentation I have ever come
acrossed. I have found two or three docs on how to install it, but they are all
so different and make huge assumption about what packages you have installed
and locations of files. Has anyone seen something better, I want to get this
w
On 3/3/06, Tomislav Parčina <[EMAIL PROTECTED]> wrote:
> In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> > Does this belong to my dialplan or my sip registration settings?
>
> To your SIP registration settings. You should limit that user/peer/friend to
> only one line.
>
How is the be
On 3/3/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
>
> On Mar 2, 2006, at 9:46 AM, Matt wrote:
>
> Doesn't it seem absurd to go through all these gyrations, rather then
> troubleshooting and fixing the problem? I know you have already tried
> without success, but this seems absurd to me.
>
> I am
Hi,
I just want to implement Music on Hold while my * tried to get the
connection on the trunks. Any clues would be welcome...
It needed for me as when failover trunk sometimes take effect, it
takes sometime for the connection and unknowingly we disconnect the
call. MOH is just to avoid this.
Th
hi
if i have an agents that figure as a member in more than one queue,
how can i login / logout him in a specific queue an not in all queues?
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hi all i have a asterisk configured and working perfectly. but i have a
problem.
if i download a softphone for example sjphone and digit for example
[EMAIL PROTECTED] i receive this call. is possible to block this?
i only want to received calls for login users...
--
.-
__
ram wrote:
Hi
how about when trying to call SIP extention to SIP extension
Local cal
even though its going to out route
when i enable SIP_IAX=YES
then its IVR in place ask 9 to dial SIP/IAX, if not its dial to
international call
How can i avoide this
check if the user belong to
Hi,
I'm looking for someone to do time-to-time mantainence on some of our
machines going up in New York. The person *MUST* be stationed in New
York.
Areas of expertise required:
- Proficiency in Linux: Slackware, Fedora
- Proficiency on Cisco Routers
If anybody is interested, please contac
I know there's a variable for the IP of a SIP channel, but I can't find
if such a variable is avaliable for a generic voip cahnnel, or at least
h323 channels (ooh323)
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In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> what are the file permissions/ownership and are they readable by the
> asterisk process ?
The problem was that wav files where in stereo mode. I have encode them and now
it works fine.
--
Tomislav Parcina
[EMAIL PROTECTED]
_
Hi all
I'm a newbie in asterisk.I install asterisk server
successfully. I configure this server to traverse NAT.
Using Xlite clients, i make a call between 2 local
networks through Internet.Asterisk server is
installed on a host with public IP. client A (in the
LAn A) and client B (in the LAN
> So, simply respawning asterisk, or checking to see if it's running
> isn't good enough, because asterisk is indeed running. We need to
> access asterisk and issue a command, and see if asterisk responds
> appropriately. If not, we can assume it has died, and we can kill it
> off (killall -9 ast
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote:
> The best way to achieve maximum manageability is to design a MySQL database
> and develop AGI scripts (in your language of choice) that work to that
> design. I've found that it has been far easier to develop complex routing
> l
I have an AGI program with an array containing a set of ${UNIQUEID}
variables for channels that may be active on the system. I need a way
for the program to tell if they are or not.
It's certainly possible using the manager interface, or appropriate
"asterisk -rx" commands, but I'd prefer to d
Anthony,
I will suggest you to use E1, you got 30 channels to communicate. I did the
integration with Toshiba CTX using E1, and no problem at all.
Asterisk as Pri_net
Toshiba as PRI_cpe
/etc/zaptel.conf
span=1,1,0,ccs,hdb3 ;this is depanding on your pbx settingyou got to
test it out, yo
[EMAIL PROTECTED] wrote:
> Looks very nice.. Is it GPL, GNU?
PBXware interface is not GPL/GNU currently.
Some time in the future we may release is it under GPL/GNU license :)...
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In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Does this belong to my dialplan or my sip registration settings?
To your SIP registration settings. You should limit that user/peer/friend to
only one line.
--
Tomislav Parcina
tparcina#lama.hr
_
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