Really, what I would do is to set up a daily restart point when there is
no or very little activity, something like running nightly:
asterisk -rx stop when convenient
and then having the monitoring script restart it immediately. Do you need
to unload the zaptel modules as well or is
Thanks for the replies.
I wonder if I could use the Yealink phone and write a connector to
Asterisk with the IAX client on Sourceforge and make the handset look
like an iaxphone? Or maybe there is some other easier solution?
All I
need is to have the ability to go
off hook/on hook,
Hi,
Can you please tell me whether Asterisk requires any audio or video codec to be
installed separately or it supports itself?
Thanking you,
Preeta
Please do not print this email unless it is absolutely necessary. Spread
environmental awareness.
The information contained in this
On Jan 28, 2008 11:12 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
Hi list,
Anyone knows where I can get information about configuring a Dialogic
card to run with Asterisk?? The model I have is D/120JCT-LS. Somebody
told me that I had to buy the driver, but I don't know if this is true
and if
Hi Dan,
Sorry to bring a thread back from the dead, but you might find the
following interesting.
Is there an easy way to achieve this with a computer generated
voice? We do not wish to manually record the messages if possible,
in the interests of a consistent message across all voicemail
It's not Asterisk, it's SIP. Transfer takes the signaling off the
Asterisk box.
In features.conf, replace blind transfer with a call to a macro. Then
redo your dialplan with the 'g' option on inward dial commands. When
the called party uses the transfer command, your macro should read the
On 29 Jan 2008, at 11:08, George Pajari wrote:
Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P
(same problem with various previous versions; same problem with
different TE120P cards).
The customer has a partial (10 B-Channel) PRI that when it is busy
(eight or more B
Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P
(same problem with various previous versions; same problem with
different TE120P cards).
The customer has a partial (10 B-Channel) PRI that when it is busy
(eight or more B channels in use), tends to fail as shown below...
SIP wrote:
We've just launched the beta of a free service which is, really, still
only JUST out of the alpha stages.
http://www.voipmagnet.com
The basic idea is this: it's an opt-in directory focused on VoIP contact
info (with elements of social networking and privacy control).
Again, the
Mark,
I thought I would chime in here on your problem. Oddly, I have having the
same issue with a PRI with similar symptoms. The odd part is that I have
never had an issue like this with a asterisk PRI setup. My setup is a PRI
with a Sangoma card with the exact same issue with 1.4.14. After a
Kevin,
After upgrading to the latest build of everything have you seen the problem
anymore?
What's your hardware and software configs? Maybe we can find a similarity in
our systems.
- Mark
On Jan 29, 2008 9:53 AM, Kevin Kiely [EMAIL PROTECTED] wrote:
Mark,
I thought I would chime in here
Hi.
We need a full featured modem bank 20+ to attend data calls.
IAXmodem only supports fax protocols because spandsp only support fax protocols.
The idea is to do a IAX wrapper like IAXmodem but with a full featured
(but propietary) softmodem library like PCTEL or linuxant.
I hate
What is the command to obtain official release notes?
On Tue, 29 Jan 2008, The Asterisk Development Team wrote:
Asterisk 1.4.18-rc2 is now available. One of the developers made a change to
chan_sip that they wanted to get in to this release. A few other bug fixes
were
added, as well.
Asterisk 1.4.18-rc2 is now available. One of the developers made a change to
chan_sip that they wanted to get in to this release. A few other bug fixes
were
added, as well.
This release candidate is published for anyone that is interested in helping to
test it for a couple of days before it
On Tuesday 29 January 2008 08:32:44 Arjan Kroon | Mobillion wrote:
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see the following output in my CLI :
Recently with Asterisk 1.4.17 I've been running into some stability issues.
I started looking through my logs, and I found this:
[Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module
'codec_g729a.so' was not compiled against a recent version of Asterisk and
may cause instability.
Ian wrote:
Just testing to see if my emails to this mailing list gets through.
Tried posting a question, but it failed
Thanks
Ian
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show agents:
6001 (First Agent) available at '6001' (musiconhold is 'default')
6002 (Test Agent) available at '6002' (musiconhold is 'default')
2 agents configured [2 online , 0 offline]
__
show queues:
Cisco routers with DSPs as ip2ip gw will do it if you want to spend a
few bucks
On Jan 29, 2008, at 2:36 PM, Khaled Chehab [EMAIL PROTECTED]
wrote:
Dears
Any one knows a standalone voip transcoder software name,not an ip
pbx.
What I want is to transcode the incoming sip calls
On 1/29/08, Richard Revels [EMAIL PROTECTED] wrote:
It's not Asterisk, it's SIP. Transfer takes the signaling off the
Asterisk box.
In features.conf, replace blind transfer with a call to a macro. Then
redo your dialplan with the 'g' option on inward dial commands. When
the called party
- Original Message
From: Richard Revels [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 12:21:16 PM
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
It's not Asterisk,
Asterisk supports a whole bunch of codecs in the regular install -
ulaw, alaw, gsm,ilbc being the more popular ones. A common paid codec
is g729 - avbl at digium.com
-rajeev
On 1/29/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
Can you please tell me whether Asterisk requires any
Dears
Any one knows a standalone voip transcoder software name,not an ip pbx.
What I want is to transcode the incoming sip calls from g711 to g723 or
ilbc or g729 . and forward it to a media gateway ..
Regards
Khaled chehab
*
No employee or
Hi,
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see the following output in my CLI :
Ignoring entry '612345678' with no = (and not
Just testing to see if my emails to this mailing list gets through.
Tried posting a question, but it failed
Thanks
Ian
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I've tried exiting the CLI in hopes that my being in there, though it
wouldn't make any sense, was keeping it from restarting. No luck.
I've already setup a cron script to restart asterisk at night when there is
no traffic going over it. But I hate to just treat the symptoms. I want to
solve the
Kevin,
After upgrading to the latest build of everything have you seen the problem
anymore?
Don't know yet, waiting for it to break ( not a good feeling as you know)
What's your hardware and software configs? Maybe we can find a similarity in
our systems.
It's a dell poweredge with
Alex Balashov wrote:
What is the command to obtain official release notes?
On Tue, 29 Jan 2008, The Asterisk Development Team wrote:
Asterisk 1.4.18-rc2 is now available. One of the developers made a change to
chan_sip that they wanted to get in to this release. A few other bug fixes
I was wondering under what conditions Asterisk will hand off a call to
another switch.
I'm trying to verify that my local PSTN's Coppercom switch operates
correctly... and wanted to know how to get a call REFER'd to another
end-point.
Thanks,
-Philip
Allen Casteran wrote:
Anyone know what the POE draw is for the Aastra 480i phones?
We have switches that will do 15 watts on 12 ports but only do 7.7 watts on
all 24 ports.
A Cisco 3560 switch will do 15.6 watts on all 24 ports.
Just trying to find out if we need that much power.
Queue members can be SIP channel names, including ones that are reachable
at remote destination URIs, if that is what you re asking. e.g.
member = SIP/[EMAIL PROTECTED]
Queue members are made persistent in AstDB with the 'persistentmembers =
yes' option and survive reboots.
On Tue,
Disbaling transfers is an attractive option from my point of view but not
from my customer's. Being able to transfer an incoming call from the
receptionist to the required person is something businesses will consider
changing provider for in my experience.
The provider can disable
Does anyone have experience using ShoreTel SIP trunks to integrate an
Asterisk system?
I am having trouble when the ShoreTel system transfers an incoming call
from a SIP trunk to the voicemail system. From the SIP traffic, it looks
like it negotiates a codec correctly, but once the RTP stream
Hey there,
I've actually looked through the site a bunch and found some great information.
The thing I'm missing, and I think it's because of my lack of experience with
Asterisk and setting up a dial plan, is the multitude of ways/places where I
can instantiate the AIG command. Do I have to
- Original Message
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 8:05:00 PM
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
Grey,
Just tested with 1.2.13
Matt [EMAIL PROTECTED] writes:
Asterisk is doing exactly as it should.. when it steps out of the media
path, the CDR is also dropped, as asterisk is no longer responsible for
that call.
Even if Asterisk stays in the media path, the CDR's are dropped.
It is an annoying problem. Hopefully the
Philip Prindeville wrote:
I'm trying to verify that my local PSTN's Coppercom switch operates
correctly... and wanted to know how to get a call REFER'd to another
end-point.
I don't think Asterisk will ever generate a REFER, but the only possible
way it could would be using the Transfer()
- Original Message
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 7:28:32 PM
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
Grey,
I don't think you
Grey,
Just tested with 1.2.13
Asterisk always (blind or attd xfer) creates 2 records.
A few points, NEVER rely on source as the billable number.
Always use account codes.
Match the lastdata field against dst fields to figure out that it was
an xfer when doing the rating. The lastdata field will
Grey,
I don't think you understand how transfers work. Let's take for example:
USER-1 dials LOCATION A and then LOCATION B (referred to as 1,A,B).
1 Dials A and transfers the call to B.
The call data is now NO LONGER in the asterisk path, therefore asterisk has
nothing to do with the CDR.
Hi List;
Anyone tried to use speex, ilbc and g729 and come back
with a preferred one in the quality?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
bilal ghayyad wrote:
Hi List;
Anyone tried to use speex, ilbc and g729 and come back
with a preferred one in the quality?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
Thanks for the response. I have not bought it yet and here are the specs I
am considering. Any comments before I make the purchase.
*PowerEdge 2950 III*
Date 1/29/2008 12:40:22 PM Central Standard Time Catalog
Number 4 Retail 04Catalog Number / Description Product Code
The issue is simple. You make the voicemail box be the same as the room
number, then you get:
Playback(Welcome-2-nursing-home)
Voicemail(xxx,s)
Voicemail plays you've reached room, xxx, please leave a message then
beeps and records message.
On Jan 7, 2008 5:29 PM, Daniel Cole [EMAIL PROTECTED]
Have a look at asterisk-java.org. I has everything you need.
On Jan 29, 2008 4:35 PM, Evan Ruff [EMAIL PROTECTED] wrote:
Hey Guys,
I've been doing some research into the AGI-Java connector and was
wondering if somebody could help me with my architecture.
What I'd like to do, is kick
Hey Guys,
I've been doing some research into the AGI-Java connector and was wondering if
somebody could help me with my architecture.
What I'd like to do, is kick off an external java class when a user:
1. Initiates an outgoing call
2. Hangs up the outgoing call
3. Has an incoming call
Hopefully a fairly easy question for the group...
I have a queue which should contain about 10 agents (it will be all the
phones in the office). This office is remote, so I would like to add
their sip phones into the queue remotely. Also, if the system ever gets
reloaded or rebooted, I need those
Hi Matt,
Sadly I understand all to well how transfers work. I've had to go over and
over this for the last 12 months trying to find different ways of handling
it. I'm talking about blind and attended call transfers here not IAX or any
other kind. We are not taking Asterisk out of the
- Original Message
From: Philip Prindeville [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 7:11:01 PM
Subject: [asterisk-users] When does Asterisk REFER?
I was wondering under what
Grey Man wrote:
That will work for blind transfers but not attended and even in the blind
transfer case the CDR's still aren't correct you're relying on an
informational field.
I think there is an important point being missed here; Asterisk did not
originate the concept of CDRs, nor did it
- Original Message
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 8:39:25 PM
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
Grey... I'm not debating
Someone could please help me ?
Regards,
Fernando
Fernando Berretta wrote:
Dear Lacy,
We are using Standard FreePbx installation and we are trying to direct
pickup all the calls with **EXT NUMBER.
[app-pickup]
include = app-pickup-custom
exten = _**.,1,Noop(Attempt to Pickup ${EXTEN:2} by
Rob Schall wrote:
Hopefully a fairly easy question for the group...
I have a queue which should contain about 10 agents (it will be all the
phones in the office). This office is remote, so I would like to add
their sip phones into the queue remotely. Also, if the system ever gets
reloaded
Hi List,
I have a scenario that I want to try out (we potential have a client who would
need this), but I am as of yet unable to find much help with it.
What we want to do is have an asterisk box with a large number of extensions
(1000+). This asterisk box will have approximately 3 SIP trunks
I would broker the dial-out requests through FastAGI and put the logic
that examines extensions and implements the load balancing / distribution
in there.
On Wed, 30 Jan 2008, Daniel Cole wrote:
Hi List,
I have a scenario that I want to try out (we potential have a client who
would need
- Original Message
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 9:34:23 PM
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
Grey Man wrote:
No, there is no need for any audio codec to be installed. In that case
it would just be the words worst B2B SIP UA.
By default Asterisk comes with quite a few codecs.
On Jan 29, 2008 5:03 AM, [EMAIL PROTECTED] wrote:
Hi,
Can you please tell me whether Asterisk requires any audio or
- Original Message
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 9:24:14 PM
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
The provider can
I think its totally possible. Most of these winmodems are just sound
cards with a hybrid.
On Jan 29, 2008 11:51 AM, Pepe Aracil [EMAIL PROTECTED] wrote:
Hi.
We need a full featured modem bank 20+ to attend data calls.
IAXmodem only supports fax protocols because spandsp only support fax
- Original Message
From: Daniel Cole [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 10:31:55 PM
Subject: [asterisk-users] Source Based Call Routing
Hi List,
I have a scenario that I
On Jan 29, 2008 3:54 PM, Grey Man [EMAIL PROTECTED] wrote:
- Original Message
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 29 January, 2008 8:05:00 PM
Subject: Re: [asterisk-users]
Franklin,
Because ChanSpy() is a passive monitor, there is nothing about the
implementation that would cause Asterisk to shunt the speech back to
itself. Asterisk only does this in situations where it is out of the
media path and needs to insinuate itself back into it for the purpose
of
Hello all,
I am allowing a reinvite between a snom 320 phone and a SIP gateway to take
load off my Asterisk server. When I put the caller on hold, for example,
Asterisk successfully reinserts itself into the rtp stream to play music on
hold to the caller, but when I do a chanspy Asterisk does
You can also look at routing based on number ranges (if you keep the
separate numbers in separate number ranges) but I would guess that this
is not going to suit your needs.
Maybe storing all the accounts in mysql (realtime) would also be a good
planh.
PaulH
On Wed, 2008-01-30 at
I would still say the easiest thing by far is to introduce a mediator
in the dial plan that is far more intelligent and extensible than the
dial plan logic itself. Enter FastAGI. Then you can just do it ...
however you want.
On Wed, 30 Jan 2008, Paul Hales wrote:
You can also look at
Thank you Greg and Alex for your contribution.
I will use your leads to see what I can get asterisk to do :)
Many Thanks,
Daniel
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man
Sent: Wednesday, 30 January 2008 9:39 AM
To: Asterisk Users
I am using Queue to handle some incoming calls. I wonder if the agent is
across multiple servers, will this work?
Thanks in advance
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Basically you initiate the call via a Originate method inn your Java App and
bridge it to your AGI. Follow the documentation on the site and you should
be in a good path.
Best Regards,
On Jan 29, 2008 7:01 PM, Evan Ruff [EMAIL PROTECTED] wrote:
Hey there,
I've actually looked through the
How to make ${ANSWEREDTIME} to work with Queue, so when the user hangs up, I
can calculate how much time the each call lasts?
Thanks
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I just save a bunch of money on my car insurance by switching to AIG!!!
Sorry, couldn't resist,
Steve Totaro
On Jan 29, 2008 8:11 PM, Guilherme Loch Waltrick Góes [EMAIL PROTECTED] wrote:
Basically you initiate the call via a Originate method inn your Java App and
bridge it to your AGI. Follow
On Jan 29, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote:
Franklin,
Because ChanSpy() is a passive monitor, there is nothing about the
implementation that would cause Asterisk to shunt the speech back to
itself. Asterisk only does this in situations where it is out of the
media path
Hello everybody
Anyone, to know a gateway that works with nextel simm cards?
I'm looking for them, in internet, but I did'n look.
Best regards
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Hi All
For the scenario of a single asterisk server that needs to serve clients
on the net, as well as local office clients, I would be very interested
in people's views of the best method to handle security to prevent net
based attacks while still allowing the client access.
Some of the
Any GSM 900/1800 gateway will work with a Nextel (US) SIM card.
However I assume you actually want to register on a local iDEN network
and not be roaming internationally (Nextel does not have any GSM
roaming partners in the US) That is not possible.
On Jan 29, 2008 9:34 PM, Carlos Rojas [EMAIL
Does anyone know the cause of these BAD BAD BAD messages?
I think I lost all my calls when it happened too. We have nagios running
against IAX and nagios reports that IAX is down. It would seem that the entire
application locks up when this happens and calls are dropped.
Connected to Asterisk
Everything seems find on my end. Here's the setup:
Linksys SPA922 - Asterisk 1.4 --- Quintum T1 gateway
Between Asterisk and Quintum if I use G729 RFC2833 DTMF works with no
issues, however if I use uLaw this is where there is a problem. For
some reason the Quintum gateway does not
It doesn't actually work at all - I tried, and even logged a bug with
digium with no luck. :(
Are the queue logs not quite good enough?
PaulH
On Tue, 2008-01-29 at 17:20 -0800, Johnny Tam wrote:
How to make ${ANSWEREDTIME} to work with Queue, so when the user hangs
up, I can calculate how
Daniel,
attach a dialplan variable to each extension using setvar
in sip.conf:
[6318]
type=friend
username=6318
secret=xx
host=dynamic
nat=no
dtmfmode=rfc2833
qualify=0
amaflags=billing
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=phone
setvar=__usetrunk=1
you can use the
Hi all
I have a small problem here. I asked this question on another asterisk
mailing list, but nobody seemed to be able to help me there.
We are running
* Asterisk 1.4.17
* Libpri 1.4.3
* Zaptel 1.4.8
on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo
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