When I use Dial(type/identifier, timeout, A(some_file))
CDR billsec starts when announcement ends. But I have to bill from when
called party answers to phone.
How can I solve my problem?
--
Suich
___
--Bandwidth and Colocation Provided by
Paul Hales a écrit :
The current Digium BRI cards need the phones to send DTMF over as
SIP-INFO.
Not sure why, but googling should help. (I think this is even covered on
the Digium site)
Hi Paul,
don't quite understand what you mean: how can a regular phone (not IP)
send SIP Info DTMF?
It could be a good idea. Right now, I'm trying to verify it's possible
to identify a caller and his postion in queue with his URI after having
got the result from the command sent to Asterisk Manager.
Scott Wolfe a écrit :
Write a application to log the information to a DB, then have all
Hi, my problem isn't on new voip box with latest asterisk version...my
problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this
steps for remove rightly TDM Card:
- remove line configuration about tdm card in zapata.conf and zaptel.conf
- remove in rc.modules and
Hi All;
I would like the needed hardware (MHz, MB, and GBI)
for the following:
1) Users: 30 IP Phones.
2) IP Trunk for maximum 10 concurrent calls, with g729
codec.
3) Analogue card of 8 lines FXO.
4) Softphone 5 and they use g729 codec.
5) Functions to be normal functions (call pickup, call
Hi,
This procedure should work whether you have remove some VoIP card from
your VoIP box.
Anyway be careful
On Nov 29, 2007 11:14 AM, Sasa [EMAIL PROTECTED] wrote:
Hi, my problem isn't on new voip box with latest asterisk version...my
problem is on voip with Asterisk 1.2.13 where I must
Darryl Dunkin wrote:
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming from the PSTN via a Zaptel gateway. I am only using the
stock wav files and have not attempted to use much else so
Hi,
We seem to be having some teething issues with a new Hylafax - happy to pay
someone to complete the installation. Please contact offlist.
Regards,
Sahil Gupta
Chief Executive Officer
VoiceValley Group of Companies
Phone: +61-7-30188403
Fax: +61-7-30188499
Russell Bryant wrote:
Ron McCarthy wrote:
Asterisk 1.4 im guessing? I did not know the Snom's worked with that,
Ill have to check it out then!
The way it is implemented in Asterisk is a bit interesting. It uses the
existing device state support (hints, BLF) to manage the buttons for shared
Hi,
Yes it make sense to have multiple registrars and to have SER acting as
a transparent proxy that forwards also REGISTER messages.
The question is: why it does not work! :-(
Regards,
Stefano
Stefano,
It is not Asterisk, It is SER (dispatcher module ?).
Why Asterisk is acting as
Moises Silva wrote:
You should not care for debug messages unless you are debugging.
I have been debugging. My IAXmodem connection.
core show codecs
Will show you format 8 is ALAW
thanks.
- Moy
On Nov 28, 2007 2:41 PM, Robert Moskowitz [EMAIL PROTECTED] wrote:
The IAX2 channel
Thanks to all the people who commented.
It sounds like I don't really need it currently, but worth experimenting
with for larger deployments.
Cheers
Alan
--
The way out is open!
http://www.theopensourcerer.com
___
--Bandwidth and Colocation
Paul Hales wrote:
But a single port E1 card with hardware echo cancellationpossible?
Yes, I would say that is definitely possible (wink wink).
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)
Hi,
I would like to implement a queue using a circular strategy, I mean,
using roundrobin or rrmemory strategies. However, I am not able to
define the order Asterisk will call the agents once a new call arrives
in the queue. Seems that Asterisk will always define its order as the
queues.conf
i am using TDM400P (clone)
Message: 22
Date: Thu, 29 Nov 2007 10:20:12 +0800
From: zhao bo [EMAIL PROTECTED]
Subject: [asterisk-users] FSK signal start after second ring
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Steve Underwood wrote:
Darryl Dunkin wrote:
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming from the PSTN via a Zaptel gateway. I am only using the
stock wav files and have not
Jeremy Mann wrote:
And they work with Asterisk/Zaptel 1.4 ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson
Sent: Wednesday, November 28, 2007 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I've also looked into this issue, and it seems that asterisk doesn't
respect the order of the members in queues.conf.
Asterisk uses a hash table internally to hold the queue members. I
guess it's fine when you have dozens of agents, but for simpler
scenarios, it's a pain not to be able to
Stefano,
if you have distributed Registrars, which will keep the user location
of registration ?
And you do not need OpenSER to fwd Register message.. Register / Proxy
/ Redirect could be totally separate entities.
By the way, if you post the SIP Register message likely someone could help you.
It's a nortel phone switch (ie: phone company), not a nortel pbx.
On Wed, Nov 28, 2007 at 08:55:09PM -0600, Jonn R Taylor wrote:
What LAN and you using? ELAN or HSP Are you trying to connect to a signaling
server? Please provide Nortel config.
Jonn
-Original Message-
From:
Can someone guide me on what package I can use to do least cost routing
in asterisk-1.4 without going through the prepaid calling card platforms.
I have tried Asterisk::LCR and LCDial without success, if more help on
either too. I will be glad.
I will be glad for good pointers.
Thanks.
OpenSER has an LCR module.
If you're interested in a commercial product, try www.transnexus.com.
On Thu, 29 Nov 2007, Goke Aruna wrote:
Can someone guide me on what package I can use to do least cost routing
in asterisk-1.4 without going through the prepaid calling card platforms.
I have
I should add that TransNexus's product comes with an OSP module for
Asterisk that allows calls to be brokered directly through Asterisk
in the dial plan logic, interfacing with the TransNexus NexSRS system.
On Thu, 29 Nov 2007, Alex Balashov wrote:
OpenSER has an LCR module.
If you're
I have problems with 1.4.14, it crash every few minutes.
The same configuration and machine in Asterisk 1.4.6 it doesn´t happend.
Is there anybody with similiar problems?
Thanks
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Hi All.
I've been experimenting with SLA on Asterisk 1.4.13 (patched up to
1.4.14).
I am using a SIP channel for my trunk line.
On the whole things are good, but I have noticed that if I misdial an
outgoing call,
i.e. I get 404 Not Found in the SIP trace, then the trunk line just
drops, rather
On Thu, Nov 29, 2007 at 11:14:12AM +0100, Sasa wrote:
Hi, my problem isn't on new voip box with latest asterisk version...my
problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this
steps for remove rightly TDM Card:
- remove line configuration about tdm card in
On Thu, 2007-11-29 at 14:28 -0300, equis software wrote:
I have problems with 1.4.14, it crash every few minutes.
The same configuration and machine in Asterisk 1.4.6 it doesn´t
happend.
Are you able to get a good backtrace from the core file generated by the
crash? Without more details,
Has anyone figured out a way to instantaneously swing over PRIs bearing
calls in progress to another media gateway without dropping them?
Obviously, this would require a DACS of some sort. But I am thinking
that it is possible to swing T1s over in a DACS without actually
causing the endpoint
Paul Hales wrote:
I also understand your stand here Kevin - there is no way you can
restrict the software running on a server out in the wild, and no way to
make sure the software they are running will not conflict in any way.
But a single port E1 card with hardware echo
You are right!
Here there is the backtrace
(gdb) bt
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
default, interpclass=0x0) at res_musiconhold.c:646
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64
Thank you!
It would be greate to have these feature set as a parameter in sip.conf
cheers
tomasz
On Nov 28, 2007 2:38 PM, Philipp Kempgen [EMAIL PROTECTED] wrote:
Tomasz Zieleniewski wrote:
How does asterisk detect the loop.
What are the criteria here.
What do I need to change in the
Hello Everyone,
I have a 2 Asterisk Servers, one in US and another in India.
Once someone from US calls, call hit US server and then is forwarded to India
which then is answered by someone.
i.e.
Caller -- US Asterisk Server -- India Asterisk Server -- Employee(India)
The Employee in India
[EMAIL PROTECTED] wrote:
Hello Everyone,
I have a 2 Asterisk Servers, one in US and another in India.
Once someone from US calls, call hit US server and then is forwarded to India
which then is answered by someone.
i.e.
Caller -- US Asterisk Server -- India Asterisk Server --
The Asterisk.org development team has released Asterisk versions 1.4.15 and
1.2.25. These releases contain two fixes for security issues.
http://downloads.digium.com/pub/asa/AST-2007-025.pdf
* This is a SQL injection vulnerability in the res_config_pgsql module.
Default installations of
Asterisk Project Security Advisory - AST-2007-025
++
| Product| Asterisk|
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I
This might be a frequently asked question, but how do new sounds get
added to the release?
I was trying to do a popcorn extension on my phone (that gives the
date and time... maybe even getting fancy and adjusting for the
caller's timezone based on country code or area code)... but
didn't have
Is it possible with asterisk to use a single button to park and retrieve a call?
e.g. Button is labelled Park 701
- If it is not in use, park the current call to 701
- If it is in use (the associated LED will be lit), pickup the call at
701 (putting the current call [if any] on hold).
A Polycom
Asterisk Project Security Advisory - AST-2007-026
++
| Product| Asterisk|
On Thursday 29 November 2007 13:29:42 Philip Prindeville wrote:
This might be a frequently asked question, but how do new sounds get
added to the release?
Patches that use new sounds have to be added to the bugtracker, and Digium
pays for sounds to be recorded in the 3 languages that we
On Thu, 29 Nov 2007 00:06:38 -0600, John Faubion
[EMAIL PROTECTED] wrote:
Many of the thin clients fit the bill nicely. I've been using MaxSpeed
MaxTerm clients lately.
Thanks for the tip. It seems like they no longer manufacture them:
http://www.neoware.com/products/hardware/
I'll look in the
Hello
I have a setup where i have 2 asterisk servers connected over the public
internet with plenty of bandwidth, NAT on one side only. If i use IAX
between the two *'s dtmf is flawless. If I use SIP, DTMF detection is around
30% or less. I have an exten to dial into and check DTMF:
exten
chan spy does the job i belive
ram
On Nov 30, 2007 7:37 AM, Jeff Adams [EMAIL PROTECTED] wrote:
I inherited an office with phones that are hosted off-site. Everything is
skinny and G729. I see that the FreeBSD asterisk port comes with a G729
codec.
I want to record everything. If I use port
On Fri, 30 Nov 2007, ram wrote:
chan spy does the job i belive
ram
On Nov 30, 2007 7:37 AM, Jeff Adams [EMAIL PROTECTED] wrote:
I inherited an office with phones that are hosted off-site. Everything is
skinny and G729. I see that the FreeBSD asterisk port comes with a G729
codec.
I want
On Mon, 26 Nov 2007 21:23:59 -0500, Adam Moffett [EMAIL PROTECTED]
wrote:
This method should work:
${IF($[${STAT(e,/tmp/${CALLTIME}.wav)} = 1]?${CALLTIME}.wav:)}
Yes indeed :-)
===
[internal]
exten = 888,1,Playback(leave_msg)
exten = 888,n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
Thanks for the tip. It seems like they no longer manufacture them:
http://www.neoware.com/products/hardware/
No, but the Neoware e140 has a PCI expansion slot, is expandable to 1GB RAM,
and still has room inside the case for a hard drive. It is available without
Win XPe starting at $339 new.
you can try Cain Abel ( to route calls) and Wireshark to record all the
calls.
On 11/29/07, Adam Moffett [EMAIL PROTECTED] wrote:
I'm pretty sure asterisk won't do that without modification. You'll
need to do packet sniffing and decode the datathere may be products
that do this, but
On Mon, 2007-11-26 at 21:23 -0500, Adam Moffett wrote:
A simpler example reveals the problem:
exten = 188,1,Noop(${STAT(e,/bin/ls)})
exten = 188,2,Noop(${STAT(e,/not/there)})
Try that and you'll find that STAT(e,/whatever) returns 1 if the file is
found and NOTHING if the
I'm pretty sure asterisk won't do that without modification. You'll
need to do packet sniffing and decode the datathere may be products
that do this, but asterisk is not it.
And we're assuming the calls are unencrypted?
I inherited an office with phones that are hosted off-site.
At 03:10 PM 11/29/2007, you wrote:
Is it possible with asterisk to use a single button to park and
retrieve a call?
I could do this with my Aastra 480i CT as the buttons can have
different meaning for different states.
Ira
___
--Bandwidth and
I inherited an office with phones that are hosted off-site. Everything is
skinny and G729. I see that the FreeBSD asterisk port comes with a G729
codec.
I want to record everything. If I use port mirroring on my switch, is it
possible to configure asterisk to record and assemble packets that it
Hi,
I would like to originate my first call from CLI.
As I'm new to this, I'm wondering if it's possible.
When I type originate from CLI, I've got this :
There are two ways to use this command. A call can be originated between
a
channel and a specific application, or between a channel and an
52 matches
Mail list logo
