It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.
How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?
Is there some circumstances where you can ask permission from the city
ahead of time?
I realize
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi All
Has anyone got an annotated SEPmac.cnf.xml they are using successfully
with the 7970 (8.0.3 Sip) and Asterisk?
The SEPmac.cnf.xml files on the wiki are not annotated and although I've
managed to upgrade the phone firmware and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Rushowr wrote:
I'm not personally sure, but if I recall correctly, the astDB is cleared
whenever the Asterisk server is stopped...
This is not correct.
Hi Doug,
Where can I find information's about maximum data that I can store
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hello all,
Wanted to toss out a question that I've been looking into for some time now
with no real results. When a variable is given a value in the dialplan, that
obviously will take up a little memory. If you're running a rather
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
If it was a .tar.gz download then you will need to reinstall.
Hi Matt!
If I upgrade to 1.2.10 and than decide to go back to some prior version, how
will I do that (using tar.gz)?
--
Tomislav Parčina
Lama Computers Split
Stinice 12,
I don't think it's a stupid question at all. Testing 911 routing is
very important, and it would suck to find out it didn't work when you
needed it to. When I tested 911 at my wife's small business (we're
on ZAP channels), I first called the non-emergency number for our
local police
On Jul 16, 2006, at 11:05 PM, voiplist wrote:
It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.
How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?
Is there some circumstances where you can ask
I call and immediately identify this as a test call.
I state the following. My Nane, and the fact that I am the PBX tech,
(engineer confuses them). I ask them to confirm my address and call back
number I provide to them. If all is OK I thank them and hang up. I do
not think it is a false call if
On Jul 16, 2006, at 11:12 PM, Tomislav Parčina wrote:
In article [EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
If it was a .tar.gz download then you will need to reinstall.
Hi Matt!
If I upgrade to 1.2.10 and than decide to go back to some prior version, how will I do that (using tar.gz)?
Thanks for your help but where is should put this bash script ,can
you guide me please
Regards
...receiving digits from IVR through
dtmf and store it on a text file
short idea:
1 IVR start
2 set(number=)
3 playback(press_digit_or_#_to_finish)
4 (pressed)
On Sun, 2006-07-16 at 23:57 -0700, Martin Joseph wrote:
I think if you keep the older source in a separate directory, you can
always cd back to it and do a make clean, make, make install.
or if you are lazy, make takes multiple targets so you could do:
make clean all install
all on one like
Dear
I want to make a billing recharge through receiving digits from IVR
through dtmf and store it on a text file ,
How can todo
that ?
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Martin Joseph wrote:
On Jul 16, 2006, at 11:12 PM, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
If it was a .tar.gz download then you will need to reinstall.
Hi Matt!
If I upgrade to 1.2.10 and than
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Julian Varanini wrote:
So I can just install it over 1.2.9? This is what I did and everything seems
to be working fine.
Yes as long as it doesn't complain there are modules which were not
compiled for the running version i.e. app_math
- --
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Martin Joseph wrote:
On Jul 16, 2006, at 9:45 PM, Abdul wrote:
Hello,
In some countries i found that they are blocking SIP port 5060
so instead of this i change to another port 1221, and its work
well. But in one country the are not blocking
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nick wrote:
Yeah a bit messy I guess. I had been hoping for a simple solution, but
knew there most likely wasn't!
The one idea I did have would be to use some kind of SIP api on the web
backend. Then bring the backend extension into a
On Mon, 2006-07-17 at 19:21 +1200, Matt Riddell (NZ) wrote:
It will sometimes tell you that there are modules inside
/var/lib/asterisk/modules which were not compiled for the version you
are compiling. If these are not asterisk-addons modules you will likely
need to remove them.
or modules
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Zeeshan Zakaria wrote:
After several hours of searching the Internet, couldn't understand how
can I
integrate Asterisk with Sphinx voice recognition system. The sphinx
software
itself I've installed on my server.
I need help from those who
I use include in an other way than you do.
i use different extensions, not the same in each includet context, maybe
that makes more sense (to you)
[apps]
include = emergency
include = cfwd
include = mailbox
[emergency]
exten = 911,1,do stuff here
[cfwd]
exten = *31,1, enable cfwd
exten =
Do you have a soft button on the IP301? I use the 501 and it works fine, you
do have to use the special asterisk code for it to work correctly. It lets
me login, logout, make the agent available/unavailable.
You can read about it at http://bugs.digium.com/view.php?id=6119
I found you must also
Have a look at this document:
http://www.snom.com/wiki/index.php/FAQs#Q:_Why_is_there_a_humming_noise_when_using_the_headset.3FMichielThanks Michiel, that was the second thing i do, phone was connected to
a well powered/connected switch.I could understand a chep headset would do that, but a 30
On Monday 17 July 2006 2:12 am, Tomislav Parčina wrote:
If those are channel variable, they should be cleared when you hang up.
Thanks for the input, but I was thinking more in terms of clearing the
variable during the call. I use temporary variables in my dialplans.
SKM
Hello everybody,
I is possible to manage multiple call parked per line
.
I mean a caller (agent) have to park more than two
call . It is possible to retrieve caller one ,two
,three, ... with a aplliction which one display the
calling parked to a PC screen or a screen phone .
Regards
Harry
Hi,
when im using only peer to peer call without any queues, im able to
dial any extension or send any digit thru dtmf durng a call. but
whenever i use queues then no phone dials any extension during a call
or a conference. i cant even hangup a call using * key. Any ideas how
this problem can be
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi people. I want to know about call forwarding. I dial *72, and a message
say me to dial the extension , I did, then the message said is forward is
UNCONDITIONLA . But when I call , it doesn't work the forwarding.
Who can help me
According to your setting, below is meaningless. Am I right?
[apps]
include = emergency
include = cfwd
include = mailbox
[emergency]
exten = 911,1,do stuff here
[cfwd]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd
exten = 911,1, do stuff2 here
exten = 911,1, do stuff3 here
[mailbox]
Title: FW: [asterisk-users] Snom 300 headset with static noise
There is a difference in the biasing circuit for the microphones in the
headsets. Unfortunately there is no standard on the market. The snom phones
190/320/360 (lets say: type A) behave different than snom 300 (type B). So
there
On 7/17/06, unplug [EMAIL PROTECTED] wrote:
According to your setting, below is meaningless. Am I right?
[apps]
include = emergency
include = cfwd
include = mailbox
[emergency]
exten = 911,1,do stuff here
[cfwd]
exten = *31,1, enable cfwd
exten = *32,1, disable cfwd
exten = 911,1, do stuff2
Hi,
I have problems to call to brazil, frome here in germany. the asterisk is
connected to the telephone system via a pri interface. I use a preselected
provider here to call out.
when I try to call a number in brazil, a mobile phone here in the germany in
the afternoon, when it is moring in
in asterisk.conf there is
"astagidir = /var/lib/asterisk/agi-bin"
it can be used for storing any scripts/programs fo *, it is suggested
for storiong AGI scripts there
example: /var/lib/asterisk/agi-bin/dtmf2text.file.sh
Thanks for
your help but where is should put this
Tomislav Parčina wrote:
Hi Doug,
Where can I find information's about maximum data that I can store in internal
* database?
According to the Wiki:
The Asterisk database uses version 1 of the Berkley DB
So, you'd need to look up the information on the Berkeley website, to
find it's
Thanks for the reply Brad.The relevant section of sip.conf was posted:[general]regcontext=sipregistrationIf you mean extensions.conf, I wasn't creating the extension in there other than for testing. RegContext correctly creates the context on registration but does not create the extension. If I
My files were almost exactly the same. We
only have 10 channels and the clid signaling was different.
We are however still getting the same
problems. I moved the box closer to the optomux (now we have 2m cable from the
optomux to the asterisk box.)
Any other ideas? We still are
Actually, for the exten 911, it flows through do stuff, then do
stuff3 instead of do stuff2. I want to implement it because I can
maintenance the dial plan easily.
Say,
My default context is [mycontext], and your default context is
[yrcontext]. We have some common contexts but not all. So I
When asterisk receives those messages you hear when calling an
unreacheable cellular phone it sends a 'connect' over the terminating
PRI line (digium TE410P), making the call seen as billed from customer's
perspective.
I don't know if this behaviour is a bug or something I can resolve with
Hi,
I want to know about the content of ast db. It is like a registry
of the asterisk to store information about register users. The
similar user register information will be stored in DB in ARA. I want
to verify that when user sends a register request and it is valid,
asterisk will capture
Did You try CLI show dialplan ?
if You set up 911 extension in 2 diffrent context and both context are
included
in third .. only one 911 will be available. 911 first loaded to asterisk
dialplan
will be valid and second will be discarded.
Loading dialplan (example below)
[mycontext] should load
This is the tact that I take, and it's never been a problem for us.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Monday, July 17, 2006 2:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hello list,
I have just tested the new Bristuff-0.3.0-Pre-1r
(released this morning) but it seems that the hangup bug isn't resolved yet. I
installed Bristuff the normal way (just run install.sh) but Asterisk still
doesn't hangup properly. Investigation of the sourcecode revealed that the
Someone here suggest to use
macro to implement my design. As I want to use ARA in my design. If
I use macro to here, ARA will be meaningless.
Yes, I suggested macros. Sorry, what is ARA?
Steve
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voiplist wrote:
It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.
How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?
Is there some circumstances where you can ask permission from the city
ahead of
Hi Guys, I need a little help in using DTMF settings. Im using SIP and
H323 channels, both are set to use dtmf=rfc2833. 2 days ago it was
working fine, it still works fine when im in conference, for example
when i use the following extension:
exten=1234,1,MeetMe(1234|X|)
by using this extension im
Moises Silva ha scritto:
AFAIK operation now in progress is a common status when you open a
socket connection. When you use blocking sockets usually you dont see
this because the connect call does not return until the connection
is done. But when using non-blocking sockets, the connect call
-- Ita erat quando hic adveni
news.rtf
Description: RTF file
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I have had the exact opposite results. I have hooked Asterisk up with
passthrough on many different systems and always initially had setup
problems which were fixed with tweaking.
Maybe Sangoma boards will give you less trouble?
Thanks,
Steve Totaro
-Original Message-
From:
Zeeshan Zakaria wrote:
How to install kernel sources?
On 7/17/06, *Dennis Gilmore* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote:
I am trying to install zaptel on dual Xeon processor but it gives
error,
saying
If you at least setup your ftp server, and point the phones to it,
they will save a copy of their contact database so that will not be
lost.
Just edit and save an entry after server is ready and it will create
the file.
No too hard to use the web browser and look at each phone to get its
This will typically happen over internet connections. If the qualify
message is lost, or takes too long the * server will stop sending
calls. This is the normal function of qualify. I find that in most
cases it is a matter of the end user saturating his connection on his
end, assuming you
Zeeshan Zakaria a écrit :
How to install kernel sources?
As asked before :
What distro are you using ?
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When asterisk receives those messages you hear when calling an
unreacheable cellular phone it sends a 'connect' over the terminating
PRI line (digium TE410P), making the call seen as billed from customer's
perspective.
I don't know if this behaviour is a bug or something I can resolve with
Our 501's upload their configs to the server by themselves... Is this uncommon? Seems to me that if you had no config on the server at all but pointed the phones there anyways, they should upload their current set of files there and then default to using that set of configs until the server is
Is your dial plan very simple, ie bypass FREEPBX
etc, to make sure no problems.
There are also debug command in the CLI:
pri debug span Enables PRI debugging on a span
pri intense debug span
Enables REALLY INTENSE PRI debugging
pri no debug span Disables PRI debugging on a span
---BeginMessage---
Identifier 0, identifier_type 2 not found in identifier list given when
sql query is :
SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\
Customer.id\ WHERE\ subscriber.username\ =\
It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686
On 7/17/06, Olivier Picquenot [EMAIL PROTECTED] wrote:
Zeeshan Zakaria a écrit : How to install kernel sources?As asked before :What distro are you using ?
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On Mon, 2006-07-17 at 15:17 +0200, olivier.taylor wrote:
email message attachment (where is the error?)
SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\
Customer.id\ WHERE\ subscriber.username\ =\
I searched these pages already, but don't understand what is needed to be done. They are missing a few steps which are needed for people not very advanced in programming.
On 7/17/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Zeeshan Zakaria wrote: After
Zeeshan Zakaria a écrit :
It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686
Then you might want to use yum to install the apropriate package, the
one that contains the kernel source, or at the very least the kernel
headers .
Or you might grab it on a Cent OS mirror, for exemple:
Callme stupid, but im not understanding your problem. Suggestions that
may help others to answer:
1. A little bit more clear in your examples? :)
2. Try describing the Asterisk behaviour under every circumstance.
Regards
On 7/17/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote:
Hi,
I have
I have dialled into a Queue, and an agent has answered the call with
AgentcallbackLogin().
The agent hits #1, to transfer the call. Asterisk responds with 'Transfer',
followed by dial tone.
As soon as I enter a digit, Asterisk responds with 'I am sorry. That is not a
valid extension'
This is
Hi list,
What is ZapRas used for ?
I would like to use asterisk as a RAS server replacing a Cisco RAS server
where users calls to a number directed to asterisk, and here, asterisk
answer the data calls and assign an IP address via PPP to calling user.
Is is possible ?
Thanks.
Hi all,
I was refreshing a running asterisk with last versions.
I am no more able to compile zaptlel package; make hung on vpm450
I saw it was introduced last 7/7/2006
(http://ftp.digium.com/pub/telephony/zaptel/releases/ChangeLog-1.2.7)
I don't know which is the purpose of this driver, but
Douglas Garstang ha scritto:
I have dialled into a Queue, and an agent has answered the call with
AgentcallbackLogin().
The agent hits #1, to transfer the call. Asterisk responds with 'Transfer',
followed by dial tone.
As soon as I enter a digit, Asterisk responds with 'I am sorry. That is
Angel Diaz wrote:
Hi list,
What is ZapRas used for ?
I would like to use asterisk as a RAS server replacing a Cisco RAS server
where users calls to a number directed to asterisk, and here, asterisk
answer the data calls and assign an IP address via PPP to calling user.
ZapRAS allows
http://bugs.digium.com/view.php?id=7536
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 17 July 2006 15:25
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] can no more compile zaptel !!!
Hi all,
I was refreshing
Ok, I found it is an open bug.
http://bugs.digium.com/view.php?id=7536
so I will follow that bug there
thanks ,
Andrea
[EMAIL PROTECTED]
.it
You can use svn export to grab a copy of the source and then archive that
directory. Roughly the same difference.
-jwb
Sent via BlackBerry from Cingular Wireless
-Original Message-
From: Matt Riddell (NZ) [EMAIL PROTECTED]
Date: Mon, 17 Jul 2006 19:21:37
To:Asterisk Users Mailing
Last week I had asked about which * version to use. The response was
that if using queues, 1.2.4 was stable and another response stated that
1.2.9 was stable with queues as long as CallBackLogin was not used.
Has this been addressed in 1.2.10? Is it even accurate or should I be
looking to
Greetings list,
I've been bashing my head against a brick wall for a couple of weeks now to
try and get this sorted, have been scouring google/the asterisk-users list
archives to no avail.
The problem I am having is that one extension (an off-site iaxy) cannot
transfer incoming calls from our
Been
working with Polycom 301/501/601 for almost a year now and I've _never_ seen
that behaviour!
I'd
love to see ngrep output of the communication between the phone and the FTP
server for this.
-Original Message-From: Alex Robar
[mailto:[EMAIL PROTECTED]Sent: Monday, July 17,
I would like to setup asterisk with Realtime and radius authentication,
but the radius patches are either outdated ( they support a version of asterisk
before realtime was mature ) or they dont patch right.
I tried this, but the version it is for is really old.
PortaOne Radius auth -
Hi Kevin, thanks for answering.
From the problem you are having it sounds like
the agent whose phone keeps ringing is in a lower penalty then the other
agent. Are both agents in the same group?
Yes, both agents are in the same group.
If you make the one agent busy
does it ring to the next
I've finally worked out how to use Asterisk assisted
transfers, from features.conf, with # and *.
Question: With an attended transfer, while the the
transferring party is announcing the original caller to the new party, the
original party does not hear music on hold. How can we enable
It's
become apparent that Asterisk does not support the ability of queue agents to
transfer callers in the queue, out of the queue. When we tried to do this, the
Queue application would completely hang. Subsequent calls into the queue would
also then hang, and the system got screwed in
After upgrading my phones I now see routine error messages:
-- Got SIP response 400 Bad Request back from 10.5.1.94
Asterisk SVN-trunk-r7230
Cisco 7960 SIP version 8-3-0.
Sip show peer:
* Name : 14012
Secret : Set
MD5Secret: Not set
Context : labcm33
Thanks, i set immediate=no and configured the incoming extensions. The ISDN
line has through selection (direct selection) and sometimes the network does
not send me the extensions and stop to the last digit of root number. Normally
i get the dialed number by ${DNID} variable, but in this case
Looks like the MWI broke on 8-3 also...
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thx mate,
but also ' must be escaped ' has to become \'
I got it, thanks for the help, u got me to the right way :)
Olivier
trixter aka Bret McDanel a crit:
On Mon, 2006-07-17 at 15:17 +0200, olivier.taylor wrote:
email message attachment (where is the error?)
I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first
preference, and the phone on the fast link to use G711 as it's first preference.
sip.conf has:
[general]
allow=ulaw
allow=g729
[slow-link] ; Override
I'm expecting regcontext to create a context of regcontext and an priority 1 extension for either the value of regexten or the peer name. The context is created, the extension says it is created but isn't. It works fine with a staticaly defined extension of the same name as defined for regexten or
I do it all the time, after I finish installing a PBX (asterisk or
other PBX) I dial 911 and say: Hi this is a test call, I'm a PBX tech,
just finished an installation and just wanted to make sure that 911
works. Then I ask the operator on the other end of the line to confirm
the e911 info he has
I have been trying to read up and understand Asterisk. I have a small office of 25 people growing to 50 and have a dedicated DSL for Asterisk and another DSL for computer use and was wondering using gsm primarily how many users I could put on the asterisk box on a single dsl. Average calls is
ted jones wrote:
I have been trying to read up and understand Asterisk. I have a small
office of 25 people growing to 50 and have a dedicated DSL for Asterisk
and another DSL for computer use and was wondering using gsm primarily
how many users I could put on the asterisk box on a single dsl.
On Montag, 17. Juli 2006 6:40 ted jones wrote:
I have been trying to read up and understand Asterisk. I have a
small office of 25 people growing to 50 and have a dedicated DSL for
Asterisk
What kind of DSL? Synchronous, Async? What speed?
and another DSL for computer use and was wondering
Well, I'm still having problems using 1.2.10 with AgentCallBackLogin:
- Local channels failing to bridge to zap chans:
(Ex: Jul 17 18:56:59 WARNING[27284]: res_features.c:1381
ast_bridge_call: Bridge failed on channels Local/[EMAIL PROTECTED],2
and Zap/72-1 )
- Zap channels shown in use but
I have a problem: when i make i call from a device h323 to sip, i have no
cdr, and i don't see cdr variables for the channnel ooh323.
Anyone can help me ??
Thanx
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The place answering the calls is generally known as the PSAP (public
safety answering point). As others noted, test calls are fine as long
as you call the non-emergency number first to let them know you're about
to do it. I'll admit I don't always call in advance though. Anyway,
calling the
sip.conf:
[2944093]
type = friend
context = one_start
username = 2944093
accountcode = 2944093
subscribecontext = one_blf
qualify = no
canreinvite = no
host = dynamic
callgroup = 1
pickupgroup = 1
dtmfmode = rfc2833
nat = no
mailbox = [EMAIL PROTECTED]
callerid = Doug 2944093
setvar = cid_agent =
We need to bill the outbound call of a blind transfer using an AGI
program. We can do this at present by:
1. Accessing ${BLINDTRANSFER}. This does not give us the user to bill
to, as users are registered on a remote SER server, but it does give us
a channel name of the form SIP/ser-random
On Wednesday 12 July 2006 00:18, Michael Workman wrote:
Well that Make me Note that I will never do Biz with you
That is if you personally vouch for Greg
I have personally done non-trivial work for Nufone on several occasions and
have always been paid promptly. I personally vouch for
Anyone know of an ATA that supports lamping the message
waiting lamp on a phone? We did an install with a bunch of Sipura 2002s.
According to the product info they have message waiting indicator support and I
took that to mean lamp support. Nope stutter tone only.
Bonus
On Jul 17, 2006, at 9:25 AM, Douglas Garstang wrote:
I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first
preference, and the phone on the fast link to use G711 as it's first
preference.
sip.conf has:
I think an easy solution for you might be along the lines of #3 but
using something like one of these devices:
http://www.command-comm.com/products.html
The ComSwitch 3.0, 5500, and 7500 are all exclusionary devices. If
you're dialing outbound through it, Asterisk won't be allowed to pick up
I have been unable to get this branch of asterisk to work properly. I
can not get any SIP phone, Polycom or X-Lite, to register with the
server. If, on the same server, I recompile and install Trunk the phones
register properly. In doing this I made no changes to the conf files at
all. I simply
The only way i figured out to fix this problem was by setting
autologoff lower than Dial timeout. This way if the agent doesn't
answer, it will log off before de Dial timeout So the next phone to
ring will be the next available agent.
Cheers,
Santiago
On 7/17/06, Delca [EMAIL PROTECTED] wrote:
-Original Message-
From: Martin Joseph [mailto:[EMAIL PROTECTED]
Sent: Monday, July 17, 2006 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec Negotiation
On Jul 17, 2006, at 9:25 AM, Douglas Garstang wrote:
I have two
I have had mixed results with Modems the pass through Asterisk. I can
recommend a solution that will always work however. We purchased an
Atlas 550 from Adtran, It 'splits' our PRIs into T1, PRI, BRI, and or
POTS. It is NOT a trivial purchase but it is a great product. We also
use it to provide
On Mon, 2006-07-17 at 11:10 -0600, Douglas Garstang wrote:
[snip]
setvar = cid_agent = 80014054 ; This should set variable cid_agent to
80014054
Did you check the samples? All the lines in the samples use:
foo=bar
You have everywhere:
foo = bar
Did you try removing all those spaces and
Olivier Picquenot wrote:
Zeeshan Zakaria a écrit :
It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686
Then you might want to use yum to install the apropriate package, the
one that contains the kernel source, or at the very least the kernel
headers .
Or you might grab it on a Cent OS
hi,
i have problem with showing actual channels
asteriskshow chanels
SIP/123456789-b6c4b2 [EMAIL PROTECTED] Up Busy()
(last 2 chars are NOT showed)
but the name of channel is longer
asterisk show channel SIP/123456789-b6c4b290
how can i get full name of channel with asterisk -rqnx ?
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Monday, July 17, 2006 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Setvar=var=val in sip.conf
On Mon, 2006-07-17 at 11:10 -0600, Douglas Garstang wrote:
[snip]
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