Dears
Do any one have an idea to make a redundant plan for asterisk ,if a call
established between two points and the server interface became down ,do we
you have an idea how to let the call established till the collie or the
caller hang-up.
Regards
Hello,
I'm wondering why asterisk is not transferring the callerid to the sip device.
Scenario as follows:
sangoma --- zaptel --- asterisk --- sip --- SIP-Device
zaptel is reporting the callerid, but in the sip packages the sip-address
shows unknown as user part, as this sip debug package
Brad Templeton wrote:
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote:
Brad Templeton wrote:
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
Unless bandwidth between the * servers is a concern, then you're better
off
Hello,
Thanks you for your reply.
The number in context test of asterisk B is 150.
exten = 15,n,Dial(OOH323/150/mypeer1);or exten =
15,n,Dial(OOH323/[EMAIL PROTECTED])
I dont know how to write the Dial parameters to say that I want to call
number 150 of test
context in
Hello,
I need your advice about H323 and asterisk! ;) Which one do you advice
me to choose H323 (only gateway mode)? ooh323? ooh323c?
Which one is the best H323 module to use with asterisk? Which one did
you choose and why?
What is your return on experience?
For more informations :
Hi all,
i have ser and asterisk on the same box with a public
ip address. When an UA behind NAT registred on SER try
to call the Voicemail or another UA registred on
Asterisk i have one way audio (caller cannot hear the
callee).
[UA/SER]--[router/nat]--[SER/Asterisk]
UA has private
Hi all,
I have problem using DTMF on Snom Phones (300, 320 and 360)
I read they use in preference out-of-band DTMF , and if the remote system
does not support it they default back to inband.
I would like to use DTMF as out of band , and I defined
dtmfmode=rfc2833
in the peer configuration.
Hi all,
My asterisk box have some peers with as host the name of a dynamic dns
resolver ex: foo.dyndns.org.
All works fine, until the host foo.dyndns.org for some reason change his
ip, asterisk didn't resolve again the new ip until a sip relolad
Actually, i use a cron with a bash script to
Asterisk can manage dynamic hostnames itseld type dnsmgr refresh in
asterisk cli . Also see /etc/asterisk/dnsmgr.conf
On 10/01/07, Ale [EMAIL PROTECTED] wrote:
Hi all,
My asterisk box have some peers with as host the name of a dynamic dns
resolver ex: foo.dyndns.org.
All works fine, until
Hi,
On 1/9/07, J. Oquendo [EMAIL PROTECTED] wrote:
Has anyone got this annoying sidecar to illuminate when users are on the
phone?
Yup, works fine.
I've tried Context: Line, still no dice. In extensions.conf I have:
exten = 4000,hint,SIP/4000,name
Make sure that the hint is not the first
I prefer h323 included in asterisk tree,
I have caller id issues with ooh323 and nobody answer to bugreports
oh323 from inaccessible network is unmaintained/death project,
incompatible with asterisk 1.4.
PJ
Michel wrote:
Hello,
I need your advice about H323 and asterisk! ;) Which one do
M.Hockings wrote:
I don't really know the name of what I want to look for but maybe
someone could tell me if it would be available.
I have a number of old analogue cell phones laying about here and I
was thinking it would be useful if I could set up a short range base
station for them
Hey users,
i've got a question about calling line id in libpri - zaptel with
switchtype q.sig. My Q.Sig partner is a Siemens F900 (HiPoint). If I
enable
span debug i see messages from type CONNECT with some kind of bit field:
Protocol Discriminator: Q.931 (8) len=87
Call Ref: len= 2
Thank you all,
we succeeded to make the fax working synchronizing the clocks.
Regards,
Jeremi
On 1/9/07, Lee Howard [EMAIL PROTECTED] wrote:
jeremij jerome wrote:
The problem is with the fax. We just want to send and receive faxes
from/to our fax machine connected to the Siemens
after the Dial has connected, I want the caller (on a SIP phone) to be
able to press keys in order to record call status. is this possible?
Have a look here :
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
applicationmap is what you are looking for
hth
Dear All,
I am facing a strange problem that I can't find any matches for while
googling, sometimes while a call initiated from asterisk to the PSTN is
answered and the answering party say the receiptionist tries to transfer
the call to someone else, the call dies, the full log shows nothing
This page should help:
http://www.voip-info.org/wiki/view/Asterisk+CentOS-4.0+Zaptel
Tzafrir Cohen
[EMAIL PROTECTED]
Here is the complete output of ztcfg:
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
It appears that none of the zaptel devices have been created. I did not
notice any errors during the make install. Does anyone have any
On Wed, Jan 10, 2007 at 08:03:07AM -0600, Chris Bullock wrote:
Here is the complete output of ztcfg:
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
It appears that none of the zaptel devices have been created. I
Jerry Glomph Black wrote:
I cannot find this file anywhere, despite thorough searching.
Certainly not in the two usual big sound tarfiles. I have a great
place for this file in my extensions.conf, no doubt.
It has not been made available for distribution, sorry.
On Jan 9, 2007, at 7:01 PM, Administrator wrote:
I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri,
1.4, and Zaptel 1.4
The Digium cards installed are TDM2400 and TE110P.
Everything was working fine until I upgraded to zaptel 1.2.12 from
1.2.9
Now when I run ztcfg I get the
Mark Coccimiglio wrote:
Marty,
Where are you paying $1000 for a 1600 series Cisco? I can get you
20% off that price on any quantity (note: Sarcasam). Its not the
1990's anymore. You can get them on eBay ($50-150) for only slightly
more then the Linksys. The performance is rock solid.
Has anyone else had any difficulty with calls Originating from the PSTN
being passed to asterisk 1.4.0 unsing a linksys SPA-3000?
I've not had enough time to track down what's happening but with 1.4.0,
When a call comes in, it is passed to asterisk and then forwarded to the
extension that
Hi all, I have to make for a client an asterisk system for process up to
250 calls between conference and normal call.
At disposition I have 4 xserver 346 with dual xeon 3.0Ghz and the client
require a failover system.
Anyone have experience for this type of solution?
Is better ultramonkey,
Hello,
we are running a Asterisk (1.2) installation with about 80 snom phones
(300,320,360).
Now have the demand for a special manager - assistant setup for a few
extensions.
Since Shared Line Appearance is not available in 1.2 I´m wondering how
to realize this...
What we need is that the
Yes, I have de same problem...I dont know if there is an error...
Regards
On 12/15/06, Miguel Paolino [EMAIL PROTECTED] wrote:
I'm using asterisk blind/attended transfer feature on a queue (also
tried with sip phones feature), and both type of transfers work fine. The
problem is that
I'd wager to say yes, it does support layer 3 routing :) That's a bit of a
redundant term (though you can route above layer 3). Depending on how many
interfaces you have on your router, you may be sending multiple vlans over a
trunk port (I'm pretty sure the 1600 series support trunk ports -- you
Hi group,
I'm trying to configure the email notification when a user leave a
voicemail, but don't work (send email notification).
I configured esmtp in my linux box, if a try to use it with command
line, it works fine. (echo Hello | sendmail [EMAIL PROTECTED] -f [EMAIL
PROTECTED]).
My
Is there a way to configure the Asterisk so that the RTP goes directly
between the Endpoints as opposed to going through the asterisk?
-Dave
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with read()),
save it into a variable to insert it into a SQL server database. But I
cannot see results into the variable, it always return NULL.
Here is a piece of the AGI.
fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
Hi Michael, in practice I think that the managers extension should default
to the assistant who can screen the call or call forward it.
Call Forward - always or Call Forward - no answer would give you the
flexability required.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Option A: Use the manager interface.
Tzafrir , Thanks,
the idea to use the manager interface is wonderful. It is really fast
and no data gets lost. I don't think 4000 Rows are a noticeable
amaount of data for a db1 database.
I coded this:
#!/usr/bin/perl
use Asterisk::Manager;
my $astman =
Thanks for the help. I was concerned because I tried once before and it
formatted my hard disk. I wanted to be sure that did not happen again.\
Bob Rawlinson
On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
Has anyone heard of a build or instructions for installing Asterisk on a
I have a group of users whos complaint about Asterisk is that the directory
application is too hard too use. (yeah, yeah, I know. For the record,
they're Calgarians) Now I'm in a pickle: I don't want to have to create a
custom directory for these guys. Anyone have any tips for making the
directory
Am Mittwoch, den 10.01.2007, 16:37 +0100 schrieb Michael Hamann:
Hello,
we are running a Asterisk (1.2) installation with about 80 snom phones
(300,320,360).
Now have the demand for a special manager - assistant setup for a few
extensions.
Since Shared Line Appearance is not available
Hello all,
We have a slight issue to resolve. We have a client who we are drafting an SLA
for the delivery of telephony services using Asterisk. Nothing extraordinary.
However, we do need a way to measure our service availability.
We currently use Nagios and Cacti to monitor server
Am Mittwoch, den 10.01.2007, 11:46 -0400 schrieb H Aranguren:
Hi group,
I'm trying to configure the email notification when a user leave a
voicemail, but don't work (send email notification).
I configured esmtp in my linux box, if a try to use it with command
line, it works fine. (echo
you could use one of the AGI libraries...
then you can just call a function to get the number.
AF.
Ralph Liebessohn wrote:
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with
read()), save it into a variable to insert it into a SQL server
database. But I cannot see
Results From cat /proc/zaptel/*
Span 1: WCTDM/0 Wildcard TDM2400P Board 1
IRQ misses: 24
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXOKS (In use)
4 WCTDM/0/3 FXOKS (In use)
5 WCTDM/0/4 FXOKS (In use)
We're currently running 1.4 r48326 - a little while before the full 1.4
release.
We are having some problems (crashes) with attended transfers and some
other stuff, and was going to move to the latest svn 1.4 as I beleive
that the attended transfer bug has been fixed.
However, I note that
Formated your hardisk... wow that is nasty, but I also cannot understand
how that could ever happen. There must be some other HW problem going on
or you got a hold of some really bad code.
What code (source or binary) and what were you doing when that happenned?
Doug
On Wed, 10 Jan 2007, Robert
When I load the asterisk 1.4 gui and log into
/asterisk/static/config/setup/install.html, it tells me No Analog ports
has been detected on your system.
I have 2 Wildcard X100P cards that are properly installed. Ztcfg shows no
problems.
I also get the following message from the asterisk console
Ok. I finally got past this. After doing all the relevant udev stuff, I ran
a make config from the zaptel sources, and got the service to install.
I'm still quiet an asterisk newbie, and defiantly a huge Linux newbie, so
thanks for the help.
-Chris
I wish had some pearl of wisdom here, but I don't. I will simply
share my sympathy.
Sounds like an ESU situation to me.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
Pavel wrote:
I prefer h323 included in asterisk tree,
I have caller id issues with ooh323 and nobody
answer to bugreports oh323 from inaccessible
network is unmaintained/death project, incompatible
with asterisk 1.4.
PJ
Response to ooh323c bugs is very slow, and patches can
take some time
Is there any documentation you guys can point us to in order to learn more
about the proper use of the Local channel? We don't currently use it. However,
while evaluating other people's billing and management systems for Asterisk, we
noticed they make extensive use of it.
Thanks,
Daniel
On Wed, Jan 10, 2007 at 09:39:47AM -0700, Administrator wrote:
Results From cat /proc/zaptel/*
Span 1: WCTDM/0 Wildcard TDM2400P Board 1
IRQ misses: 24
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXOKS (In use)
4
he is probably tried to install one of these All in one Asterisk CDs, that
effectively formats the hard drive and installs everything from scratch,
including the OS ;)
And, yes, it will happen again, if he re-runs this CD...
AF.
Doug Crompton wrote:
Formated your hardisk... wow that is
Greetings,
I've been having a large number of deadlock issues lately on chan_sip occurring
only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar
issues.
My Config (have multiple systems all running the same hardware with the same
problem)
VIA EPIA ML6000
1GB RAM
80GB
From: Anton Frolov [EMAIL PROTECTED]
you could use one of the AGI libraries...
then you can just call a function to get the number.
AF.
Ralph Liebessohn wrote:
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with
read()), save it into a variable to insert it into a SQL
Thanks for your answer Anselm,
But, why do you think that the problem is in the mail server, if I
can send mails with esmtp, with the command /usr/sbin/sendmail without
problem. But the Voicemail app never sends the notification.
I used ethereal and i couldn't see any message from asterisk
hi list,
i have the same problem as mentioned here:
http://forums.digium.com/viewtopic.php?t=2678view=nextsid=bd94cefd823b23156c5748843febb3ab
my asterisk version is 1.2.12.1
any ideas?
___
Der frühe Vogel
More like a ID-10-T error.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan M.
Johns
Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory
Mark,
Do these 1600 series Cisco routers you mention that you find on eBay
for $50-$150 support Layer3 routing? I have a managed switch setup on
my home network with several VLANs defined. (work subnet, home subnet,
VOIP subnet) I currently have to use a Linux box to route between
the
It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.
James
-Original
On Wed, Jan 10, 2007 at 01:41:39PM -0400, H Aranguren wrote:
Thanks for your answer Anselm,
But, why do you think that the problem is in the mail server, if I
can send mails with esmtp, with the command /usr/sbin/sendmail without
problem. But the Voicemail app never sends the notification.
Am Mittwoch, den 10.01.2007, 13:41 -0400 schrieb H Aranguren:
Thanks for your answer Anselm,
But, why do you think that the problem is in the mail server, if I
can send mails with esmtp, with the command /usr/sbin/sendmail without
problem. But the Voicemail app never sends the
Yes you are correct. I do NOT plan to use it again. I have downloaded
the latest version and plan to do an install. I was hoping there might
be an rpm for it but does not seem to be. Thanks all.
Bob Rawlinson
On Wed, 2007-01-10 at 18:25 +0100, Anton Frolov wrote:
he is probably tried to install
Hi Ramon,
Please post your peer details from dundi.conf so we can see what your setup
is.
Also, have you tried regenerating your keys? I wound up generating my keys
twice, they just didn't work the first time, I'm not sure why.
Alex
On 1/10/07, Ramon Schönborn [EMAIL PROTECTED] wrote:
hi
I have 8 Zap channels, 25-32, all of which have their own line.
My zapata.conf file looks similar to:
group=1
context=context_1
signalling=fxs_ks
channel = 25
group=2
context=context_2
signalling=fxs_ks
channel = 26
and so forth
Quoting Mark Coccimiglio [EMAIL PROTECTED]:
Mark,
Do these 1600 series Cisco routers you mention that you find on eBay
for $50-$150 support Layer3 routing? I have a managed switch setup
on my home network with several VLANs defined. (work subnet, home
subnet, VOIP subnet) I currently
Ralph Liebessohn wrote:
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with
read()), save it into a variable to insert it into a SQL server
database. But I cannot see results into the variable, it always return
NULL.
Here is a piece of the AGI.
fwrite(STDOUT,exec Read
On 1/10/07, Yuan LIU [EMAIL PROTECTED] wrote:
Then there must be an error somewhere. The variable READ() in Asterisk
should be usable. Should be able to use SayDigits() to play it back - or
no
value is read.
Yuan Liu
Hi Yuan and Anton,
Let's put here all AGI for test:
#!/usr/bin/php -q
Administrator a écrit :
It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.
On Wed, Jan 10, 2007 at 01:25:31PM -0500, Robert A. Rawlinson wrote:
Yes you are correct. I do NOT plan to use it again. I have downloaded
the latest version and plan to do an install. I was hoping there might
be an rpm for it but does not seem to be. Thanks all.
Bob Rawlinson
suse 10.1
I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
having calls dropped. Sometimes you can stay on the phone for a long
time and sometimes the call is dropped within a minute.
There are 9 lines connected to 3 TDM04B cards. The Panasonic Pbx we
replaced did not
I'm having a bit of trouble setting up my sip.conf entries to accept
calls from a particular provider, and the problem really is that I am
unclear exactly what parts of the INVITE are supposed to match what
parts of sip.conf.
I couldn't find this info on the wiki, so if someone here can shed
some
There is certainly an rpm. Not sure about 1.4, but at least for 1.2.
AF.
Robert A. Rawlinson wrote:
Yes you are correct. I do NOT plan to use it again. I have downloaded
the latest version and plan to do an install. I was hoping there might
be an rpm for it but does not seem to be. Thanks
hi
I never really programmed in PHP, I use Perl for my purposes.
I found a good AGI library for Perl and is happy with it. It allows me to call
functions instead of parsing the input.
While looking for my library, I saw at least one for PHP. So why not to use it?
In Perl it looks like:
my
[EMAIL PROTECTED] wrote:
Is there any documentation you guys can point us to in order to learn more
about the proper use of the Local channel? We don't currently use it. However,
while evaluating other people's billing and management systems for Asterisk, we
noticed they make extensive use of
David Alcott wrote:
Is there a way to configure the Asterisk so that the RTP goes directly
between the Endpoints as opposed to going through the asterisk?
That is the default if Asterisk believes it will work. Things that
might not make it work is tTwW options to Dial, protocol transation
On 1/10/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Ralph Liebessohn wrote:
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with
read()), save it into a variable to insert it into a SQL server
database. But I cannot see results into the variable, it always return
NULL.
Here
Strange! I had checked on both my DVD and on the Suse site and I have
not been able to find it. Do you happen to know where it is located?
Bob Rawlinson
On Wed, 2007-01-10 at 21:03 +0200, Tzafrir Cohen wrote:
On Wed, Jan 10, 2007 at 01:25:31PM -0500, Robert A. Rawlinson wrote:
Yes you are
Hi!
On Wed, Jan 10, 2007 at 01:14:59PM -0600, Carlos Chavez wrote:
I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
having calls dropped. Sometimes you can stay on the phone for a long
time and sometimes the call is dropped within a minute.
There are 9 lines
Could you point me to where it is located? I had tried Suse and
sourceforge.
Bob Rawlinson
On Wed, 2007-01-10 at 20:18 +0100, Anton Frolov wrote:
There is certainly an rpm. Not sure about 1.4, but at least for 1.2.
AF.
Robert A. Rawlinson wrote:
Yes you are correct. I do NOT plan to
In this case I would need to purchase an E1 card for the Avay PBX an an other
for the *. To save costs, I would like to intent the interconnection over the
FXO port.
Anyone has done this configuration so far?
Robert Boardman [EMAIL PROTECTED] wrote:
Just done this for a client using
Hello,
I will expose my problem here. Please tell me if it is not the right
place as I am really new to that list.
I use Asterisk as a SIP proxy. I have two users connected to it,
Let's call them 1234 and 5678
In my dialplan I have two lines:
exten = 1234,1,Dial(SIP/1234)
exten =
Exactly.
ESU = Equipment Superior to Users
;-)
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote:
More
Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it. Sure, they are built to last but they do not last
forever. I would consider ANY of these boxes as somewhat unreliable for
high
On Wed, Jan 10, 2007 at 02:48:41PM -0500, Robert A. Rawlinson wrote:
Strange! I had checked on both my DVD and on the Suse site and I have
not been able to find it. Do you happen to know where it is located?
Bob Rawlinson
I simply checked the list of source RPMs availble from the first suse
From: Ralph Liebessohn [EMAIL PROTECTED]
Hi Yuan and Anton,
Let's put here all AGI for test:
#!/usr/bin/php -q
?php
...
$my_var=123;
fflush(STDERR);
fwrite(STDERR,Just testing\\\n);
fflush(STDERR);
fwrite(STDOUT,exec read
well, I'm not rpm user anymore for several years already... Isn't it
http://www.rpmfind.com/ that is used to find the rpms?
AF.
Robert A. Rawlinson wrote:
Could you point me to where it is located? I had tried Suse and
sourceforge.
Bob Rawlinson
On Wed, 2007-01-10 at 20:18 +0100, Anton
here it is, mainly for suse
http://rpmseek.com/rpm-pl/asterisk.html?hl=comcs=asterisk:PN:0:0:0:0
it's only one of the rpms (the basic one). You should make the search yourself
(try asterisk) to locate all of them.
AF.
Robert A. Rawlinson wrote:
Could you point me to where it is located? I
On Wed, Jan 10, 2007 at 09:43:01PM +0100, Anton Frolov wrote:
well, I'm not rpm user anymore for several years already... Isn't it
http://www.rpmfind.com/ that is used to find the rpms?
It's meant to find rpm pckages not from your distribution that are not
supported and may be incompatible
No, I haven't. I'll start there.
Thanks
On Wed, January 10, 2007 2:38 pm, Eric \ManxPower\ Wieling [EMAIL
PROTECTED] said:
[EMAIL PROTECTED] wrote:
Is there any documentation you guys can point us to in order to learn more
about
the proper use of the Local channel? We don't currently use
How did you, or do go about reversing the patch?
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: Wednesday, January 10, 2007 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
shadowym wrote:
Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it. Sure, they are built to last but they do not last
forever. I would consider ANY of these boxes as somewhat
I have a DID vendor that wants me to be able to generate specific SIP
error messages under certain conditions and I'm completely stumped on
how to do these:
#1 - They want to see a SIP 503 error response(service unavailable) when
they send the call in to an active extension and and the
Hi Folks,
I implemented an Asterisk 1.2.10 on a Debian GNU/Linux, and I have
implemented a round robin queue (and a memory round robin queue too).
Here I have one simple problem:
- agent 1 (busy)
- agent 2 (busy)
- agent 3 (free)
When someone call to my queue, the action of the queue is this:
Quoting Mark Coccimiglio [EMAIL PROTECTED]:
shadowym wrote:
Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it. Sure, they are built to last but they do not last
forever. I
From: Tobias Unsleber [EMAIL PROTECTED]
Hello,
I'm wondering why asterisk is not transferring the callerid to the sip
device.
Scenario as follows:
sangoma --- zaptel --- asterisk --- sip --- SIP-Device
zaptel is reporting the callerid, but in the sip packages the sip-address
shows unknown
Ralph Liebessohn wrote:
Hi Lee,
thanks for the tip. I tried other methods trying to get the variable
value, but no success.
Doing a GET VARIABLE my_var after READ the get variable returns the
value I dialed, but doesn't give the exact value to it. I got Resource
ID #1 instead.
Using:
Ralph Liebessohn wrote:
Using:
fwrite(STDOUT,exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n);
fwrite(STDOUT,get variable my_var \n);
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,exec saydigits $my_var \n);
I got it:
Also you might try concatenating
Hi,
How did you, or do go about reversing the patch?
I have put the patch (simple) available at :
http://www.scopserv.com/download/patches/zaptel-1.2.12-reverse7860.patch
Go on your zaptel src directory and do :
patch -p0 zaptel-1.2.12-reverse7860.patch
It is a T1 and I am not sure
I got a requirement list just now, with my comments inline: (showing it just
for a giggle)
User requirement: 1) Directory set up by name - If person calling does not
know employee's name, how will they access?
-Why, using app_telepathy.so of course!
User requirement: 2) Directory set by
but some of the packages are labeled to be for SuSe 10.1 ...
AF.
Tzafrir Cohen wrote:
On Wed, Jan 10, 2007 at 09:43:01PM +0100, Anton Frolov wrote:
well, I'm not rpm user anymore for several years already... Isn't it
http://www.rpmfind.com/ that is used to find the rpms?
It's meant to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
What is needed is a family of astdb manipulation commands:
astdbput family key value
astdbget family [key]
astdbdel family [key]
any others?
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No.
Jon Pounder wrote:
you should take your own advice - an acre is 200ft x 200ft - what
idiot would
pay a consultant $7000 to tell them they need one access point in the
middle.
I have a BA in Electronic Engineering, a Masters in Computer Science and
I'm an FCC licensed
radio operator.
On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
Has anyone heard of a build or instructions for installing Asterisk on a
Suse 10.1 system?
Bob Rawlinson
Hi Bob,
Afair, asterisk was not on the cdrom's (which some people use to make
their own dvd), but it was on the original
Dumpolid Exeplish wrote:
It is true what Eric and Steve have said, you do need a licensed GSM
frequency to operate and sell GSM services (even for rural areas).
however, this link might be of interest to you
http://rfdesign.com/mag/radio_field_trials_allsoftware/
That is more what I was
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