When anyone leaves a voicemail message and email notifications are enabled
it causes the cpu to go to consume 100% cpu indefinetly. Note that when
email notifications are not enabled, the issue is resolved. I have been able
to re-create the circumstances on every Asterisk
Hello again!
Thanks Sven, for your private advice. My question now is: If the Fritz is
going out of bussiness: Which card to purchase for the future?
I have a very small budget, a simple european ISDN line (three numbers, two
similar channels?). I want to use it for asterisk.
Features
Currently being frustrated trying to compile the latest mISDN and after
getting zero feedback from the isdn4linux list when I asked about it, so
wondering whether to jump ship to bristuff... (All HFC based cards)
mISDN has worked very well for me so-far - I have PBXs with 100+ days up
time
On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED] wrote:
On Sunday 07 September 2008 21:49, Atis Lezdins wrote:
On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter [EMAIL PROTECTED]
wrote:
is not work for periodic-announce-frequency and periodic-announce.
An reload is necessary.
On Mon, Sep 8, 2008 at 11:39 AM, bala krishnan [EMAIL PROTECTED] wrote:
Hi,
To disallow the native bridge between the zap channels, i enabled the t
flag in the Dial application. But i dont want to allow the callee/caller to
transfer the call.
Why would you need this? It should just take
Gordon Henderson schrieb:
So comments, ponderings or anecdotes, etc. ... ?
Bristuff worked perfectly fine for me for about 5 years.
HOWEVER, you should keep in mind, that bristuff are very extensive
patches against the zaptel dirvers and also against the core. So
regarding updates you are
On 7 Sep 2008, at 21:34, Edgar Guadamuz wrote:
Hello,
I have been testing a trunk IAX and another SIP, using sipp to
generate SIP calls to a Asterisk box.
The testing dialplan just connects to another Asterisk box, who
answers the call and playback some files.
I noticed that the cpu
Hi,
To disallow the native bridge between the zap channels, i enabled the t flag
in the Dial application. But i dont want to allow the callee/caller to transfer
the call.
Could you please tell me if any configuration needs to set in such a way
to disallow the native bridge?
I am
Hi all,
I have a trixbox2.6.1 on my one server,
i have configured sangoma A200/Remora FXO/FXS Analog AFT card on that
server,
from my zap line the incoming faxes are coming, i have setup the did for zap
channel.
my question is when i am getting any faxes, asterisk shows me rxfax
execution and
On Sun, Sep 7, 2008 at 9:57 AM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 08:24, Sun 07 Sep 08, Steve Totaro wrote:
It may be simpler to get working but will it be simpler to
diagnose the audio issues that will invariably come down the pipe?
How about the rather popular error I should
Hello!
I'm wondering which is the best choice (kernel version and mISDN) to get my
AVM Fritz A1 PCI card to work properly?
Does anyone have an AVM Fritz runningunder Linux? Or has anyone deep
knowledge of mISDN? Please I need some hellp here, for after a whole lot of
testing, reading,
2008/9/8 Stefan Gofferje [EMAIL PROTECTED]
Gordon Henderson schrieb:
So comments, ponderings or anecdotes, etc. ... ?
Unfortunately, I must say we had bitter experiences recently with both
mISDN and Bristuff : echo, CLI, available channels tagged as busy (see
Bristuff mailing list).
On 8 Sep 2008, at 13:12, Steve Totaro wrote:
On Sun, Sep 7, 2008 at 9:57 AM, Michiel van Baak
[EMAIL PROTECTED] wrote:
On 08:24, Sun 07 Sep 08, Steve Totaro wrote:
Maybe the problem is that IAX2 is not as set in stone as the RFCs for
SIP? Who is to say it is or isn't compliant to the
Following setup :
Users are creeated in the sippers table with following Fields set
Name : .unique
Host : dynamic
Nat : yes
Type: friend
Callerid: .unique value
Context: autocreate
Secret : xx
Disallow: all
Allow : all
Username : unique : same as Name
Does/Will asterisk support video streaming on hold?
Been playing with videphones as of late, and a client asked about video on
hold - standard MoH works fine - but on the target video phone the image
just freezes - any way to inject a video?
Cheers,
Gordon
2008/9/8 Max Alex [EMAIL PROTECTED]:
Hi all,
I have a trixbox2.6.1 on my one server,
i have configured sangoma A200/Remora FXO/FXS Analog AFT card on that
server,
from my zap line the incoming faxes are coming, i have setup the did for zap
channel.
my question is when i am getting any
Patrick Maartense schrieb:
Users are creeated in the sippers table with following Fields set
Name : .unique
Host : dynamic
Nat : yes
Type: friend
Callerid: .unique value
Context: autocreate
Secret : xx
Disallow: all
Allow : all
Username :
On Mon, Sep 08, 2008 at 04:28:52PM +0200, Philipp Kempgen wrote:
Apart from that I'd appreciate if you could get a better email
client which does not insert so many useless blank lines. :-) SCNR.
Likely, for you, like me, it's not that his email client is indersting
blank lines... it's that
Off course the columns all are fine ( right case) ( to much german language
makes one write words with uppercase almost every word :(
There seems to be an irregularity between the sip peers table and the Sip
registry.
For the Client. Well you know, one that has the installed base, also set the
Steve Davies schrieb:
If you build rxfax against one version of libspandsp, and try to run
it against a different version, I have seen the crash you are
describing. It may even be that you have old .so files on your system
that you are unaware of.
That's why the naming convention for shared
Eric,
So what is practical for a PRI then?
Thank you!!!
++Amaru
--- On Fri, 9/5/08, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
From: Eric ManxPower Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] FAX over T1 Question
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
I don't know what Eric is talking about. My advice applies to T1 PRI
as does the rest as far as I care to read.
Thanks,
Steve Totaro
On Mon, Sep 8, 2008 at 10:56 AM, Amaru Netapshaak
[EMAIL PROTECTED] wrote:
Eric,
So what is practical for a PRI then?
Thank you!!!
++Amaru
--- On Fri,
Hello,
Someone has worked with the astmanproxy? I am stating the use
astmanproxy with the AutoFilter ON, but it does not filter all events, I
am doing the test and he is still receiving some events to other
channels.
He managed to filter all events? Or only some?
Below are the
Hello
I try to login in asterisk using java - manager api.
extension.conf:
exten = 100,1,Agi(agi://localhost/AgentLogin.agi)
I would call to this number and login in Asterisk. Is this possible.
for example:
originateAction = new OriginateAction();
Jay R. Ashworth schrieb:
On Mon, Sep 08, 2008 at 04:28:52PM +0200, Philipp Kempgen wrote:
Apart from that I'd appreciate if you could get a better email
client which does not insert so many useless blank lines. :-) SCNR.
Likely, for you, like me, it's not that his email client is indersting
On Mon, Sep 08, 2008 at 04:51:21PM +0200, Philipp Kempgen wrote:
Steve Davies schrieb:
If you build rxfax against one version of libspandsp, and try to run
it against a different version, I have seen the crash you are
describing. It may even be that you have old .so files on your system
On a PRI calls come in on ANY B-channel. Therefore you cannot just
disable EC on the Fax channels, because there are no dedicated channels
for fax. On a Channelized T-1 you can dedicate channels for fax or any
other thing. You can't do that on PRI.
Steve Totaro wrote:
I don't know what
Hello everyone.
What I'm doing:
I've made a replacement for app_queue that uses MeetMe to connect the
calling party with the agents. When the call comes in it gets put into a
MeetMe room with a nice AGI_BACKGROUND so the calling party can listen
to music and announcements until an agent
On Mon, Sep 08, 2008 at 10:54:12AM -0500, Eric ManxPower Wieling wrote:
On a PRI calls come in on ANY B-channel. Therefore you cannot just
disable EC on the Fax channels, because there are no dedicated channels
for fax. On a Channelized T-1 you can dedicate channels for fax or any
other
On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote:
Let me clarify some of this.
Under no circumstances can Asterisk receive a TBCT request. We just
ignore them. We can initiate them however.
There are different TBCT implementations, dependent on which switch type
is
2008/9/8 Rodrigo Pinto [EMAIL PROTECTED]:
Hello,
Someone has worked with the astmanproxy? I am stating the use
astmanproxy with the AutoFilter ON, but it does not filter all events, I
am doing the test and he is still receiving some events to other
channels.
He managed to filter all
context from-pri {
_8505 = {
Wait(1);
Answer();
SetTransferCapability(3K1AUDIO);
Set(GROUP(ZAP)=incoming);
Set(CDR(accountcode)=fax);
Set(CDR(userfield)=bedford);
Jay R. Ashworth wrote:
On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote:
Let me clarify some of this.
Under no circumstances can Asterisk receive a TBCT request. We just
ignore them. We can initiate them however.
There are different TBCT implementations, dependent on
Likely, for you, like me, it's not that his email client is indersting
blank lines... it's that whatever you're using to render his HTML
email
into text is doing it -- for me, it's lynx under Mutt.
No. I configured my email client not to render the text/html part.
(Not wanting to render a
Generic question,
Is there a way to detect a fax call without actually taking it as a fax
call? In a non-universal manner?
In other words, if fax tones are detected on the incoming call (on a
Sangoma PRI card for example), I'd like to transfer that call back out
of the PRI to a dedicated FAX
Hello!
Sorry, I'm sure it's stupid. but I've got a simple ISDN line and a simple
ISDN-card, now finally running. :-)
I'm using application Jack and asterisk (CLI) only to do my bidding. Now I
can make calls. But how ca I setup my extensions.conf to receive a call? I've
had an example like
On zapata.conf:
faxdetect=incoming
The detected fax calls will be redirected to the 'fax' extension on the context
set to the group of channels.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
That us a bit like wanting to know what the person calling you wants
to talk about without picking up the phone..
On 8 Sep 2008, at 17:42, JD wrote:
Generic question,
Is there a way to detect a fax call without actually taking it as a
fax
call? In a non-universal manner?
In other
Hi listers,
We want to implement one call center with asterisk. The idea is it should be
scalable, with openser as an dispatcher and bunch of asterisk servers to do
ACD, Queues, Agents things... Easy to say :(
Look closely to the current asterisk, we do see some problem:
- SIP registrations was
Patrick Maartense schrieb:
Don't bash MS,
I used to bash MS simply because it was MS, but I stopped wasting
my time. Now I bash not following email netiquette (RFC 1855 etc.)[1]
and in my experience 99 % of all mail produced by Outlook / Exchange
/ Lotus Notes fails to comply. Which leads me to
Bryce,
I have modified the settings on my Rhino CB according to your message..
It will take a day or so for me to be able to verify if that has helped or not.
My current settings (now) for all FXS channels is as follows:
Mode: LOOP
PwrDn: 0 sec
I-Loop: 25.0ms
Tx: -2db
Rx: -2db
Ring: 20
On: 20
I have an Asterisk server running iptables with a public interface
and an internal interface. I had to change the subnet of the internal
interface and now I see messages scrolling destroying.. 192.168.100.1
which is the old of the internal interface?
Sometimes outside calls are ringing busy and
Eric,
You are right, at least, in my case. My PRI gives me the inbound call on
whatever channel is available.
I have now modified my Rhino CB with the Options stated earlier in a
message by Bryce Chidester of Rhino Equipment. I now modified my
wanpipe1.conf (port connected to PRI) and set
On Tue, 9 Sep 2008, Nguyen wrote:
Hi listers,
We want to implement one call center with asterisk. The idea is it should be
scalable, with openser as an dispatcher and bunch of asterisk servers to do
ACD, Queues, Agents things... Easy to say :(
Look closely to the current asterisk, we do
Patrick Maartense schrieb:
Off course the columns all are fine ( right case)
Ok.
There seems to be an irregularity between the sip peers table and the Sip
registry.
But any idea on the backgrounds of this behaviour??
The SQL query tries to update the ipaddr, port and regseconds fields
No, but you can use a dedicated DID, but that will NOT detect the
tone, it will simply ASSume that the caller is trying to send a fax.
On Mon, Sep 8, 2008 at 12:42 PM, JD [EMAIL PROTECTED] wrote:
Generic question,
Is there a way to detect a fax call without actually taking it as a fax
call?
Hi All,
I have Asterisk 1.4.21.1 and Dialogic DMG 2000 firmware 6.0.103.
Trying to pass t38 fax calls setup like this:
fax machineATA (linksys and mediatrix)SIPAsteriskSIPDMG GWPRI
From the fax outbound through the PRI works great, seems to be
reliable, 40+ faxes with 5 different fax machines
I knew of this, but mostly ignored it since the zapata.conf method is a
universal function. Either a channel (or channel group) does fax
detection or it doesn't. I can't change it in the ongoing script.
In other words, if a call is going to a DID that _shouldn't_ do faxing,
it goes to the fax
I have a setup with a SIP DID inbound, and several SIP phones inside.
Obviously if the SIP phones are off/unplugged/otherwise not available,
incoming calls ring busy. My extensions.conf looks like this for inbound
calls:
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
So what
You're joking, right?
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,Voicemail([EMAIL PROTECTED])
Use whatever voice mailbox and voicemail context you want.
Joseph L. Casale wrote:
I have a setup with a SIP DID inbound, and several SIP phones inside.
On Monday 08 September 2008 13:11:38 Joseph L. Casale wrote:
I have an Asterisk server running iptables with a public interface
and an internal interface. I had to change the subnet of the internal
interface and now I see messages scrolling destroying.. 192.168.100.1
which is the old of the
Hy Guys!
I have Trixbox (2.6.1) set up with 2 analog ph lines going to 2 FXO
ports (2-X100P cards) I also have to deal with Panasonic hardware that
handles the initial calls. I would like Asterisk to serve as an auto
attendant for the first call and as calls come in pass them to the
Panasonic. My
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,Voicemail([EMAIL PROTECTED])
Use whatever voice mailbox and voicemail context you want.
Well, its not advancing when *no* phones are online, just ringing busy.
It does however step through just fine when they
Not really, I have all these files, I just mentioned the fields that are filled
by my queries ( nex time I will post the complete create statements )
But I think (HOPE) I found the problem
I only had SIPPEERS defined in the config
NOT sipusers
Now I have not seen these problems anymore (at
Check your bindaddr in sip.conf. Also check to ensure that you've restarted
Asterisk since changing the subnet. There are more than a few places that
we cache network information for speed purposes, and restarting the process
will fix that.
--
Tilghman
Got it, thanks!
jlc
JR,
On Mon, Sep 8, 2008 at 3:08 PM, JR Richardson [EMAIL PROTECTED] wrote:
The DMG invite sends to asterisk:
m=audio 49016 RTP/AVP 0 101 [notice the m=audio]
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 0 udptl t38
Joseph L. Casale wrote:
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,Voicemail([EMAIL PROTECTED])
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,NoOP(Dial Status: ${DIALSTATUS})
exten =
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,NoOP(Dial Status: ${DIALSTATUS})
exten = _1xx,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _1xx,n,Gosub(s-${DIALSTATUS},s,1)
[s-BUSY]
exten = s,1,Voicemail([EMAIL PROTECTED]|b)
Doug,
The Asterisk development team is pleased to announce new releases of
Asterisk and DAHDI.
For more information on the reasoning behind the transition to DAHDI,
please see the following post:
http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/
The list of packages released
Hi,
I'm looking for a GUI like ARI by LittleJohn Consulting (which is not being
maintained actively anymore, but FreePBX seems to include it) so users can
login, check cdrs, recordings, call forward, etc.
Does anyone know of any such working app that can be integrated into vanilla
Hi All,
When using a Digium card with a hardware echo cancellation module
installed, is the only thing required to enable it, is to set
echocancel=yes in zapata.conf?
I think I remember seeing somewhere that you need to make a change in
the zaptel Makefile to allow the echo cancellation to
I'm looking for a GUI like ARI by LittleJohn Consulting (which is not being
maintained actively anymore, but FreePBX seems to include it) so users can
login, check cdrs, recordings, call forward, etc.
What's wrong with just using FreePBX directly?
On Monday 08 September 2008 12:43:53 Nguyen wrote:
Hi listers,
We want to implement one call center with asterisk. The idea is it should
be scalable, with openser as an dispatcher and bunch of asterisk servers to
do ACD, Queues, Agents things... Easy to say :(
Look closely to the current
Because that would mean changing the entire vanilla framework with over 200 users on it.
Original Message
Subject: Re: [asterisk-users] OT: ARI
From: "David Backeberg" [EMAIL PROTECTED]
Date: Mon, September 08, 2008 8:00 pm
To: "Asterisk Users Mailing List - Non-Commercial
On Sep 8, 2008, at 9:15 AM, Gordon Henderson wrote:
Does/Will asterisk support video streaming on hold?
Been playing with videphones as of late, and a client asked about
video on
hold - standard MoH works fine - but on the target video phone the
image
just freezes - any way to inject a
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 08:39:16PM -0400, sean darcy wrote:
sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
What messages do you get when you run in the CLI:
dahdi restart
dahdi restart
On Sep 8, 2008, at 7:31 PM, Russell Bryant wrote:
On Sep 8, 2008, at 9:15 AM, Gordon Henderson wrote:
Does/Will asterisk support video streaming on hold?
Been playing with videphones as of late, and a client asked about
video on
hold - standard MoH works fine - but on the target video
Because that would mean changing the entire vanilla framework with over 200
users on it.
I don't know anything about your vanilla framework, or how complicated
it is. That said, there are lots of places in FreePBX where you can
interject regular vanilla dialplan, you just end up using files
On Sep 5, 2008, at 10:06 AM, Tim Panton wrote:
On 5 Sep 2008, at 15:50, Steve Murphy wrote:
Not in 1.4, but in trunk,(and 1.6.x) there is a the Bridge manager
command you can call via the manager interface, which takes two
required args, the names of the two channels to bridge, and an
David,Please don't mind me needing a GUI just for one purpose of vanilla Asterisk. I do have a separate box, which actually runs PBXiaF and I like it a lot.However, when a project that big in vanilla has been going on, moving on to something else makes no sense, atleast for now.So, either you
ARI really only let people check their voicemail via a web interface -
for CDR's you can install areske cdr interface as that bolts on to vanilla
asterisk with a small amount of work.
Recordings - how complicated an interface do you need? From memory
there's something in the contribs folder
On Mon, 8 Sep 2008, Tilghman Lesher wrote:
On Monday 08 September 2008 12:43:53 Nguyen wrote:
Hi listers,
We want to implement one call center with asterisk. The idea is it should
be scalable, with openser as an dispatcher and bunch of asterisk servers to
do ACD, Queues, Agents things...
Has anyone found an outlet to purchase PARTS for Aastra (or Polycom for
that matter) phones? My current example is that I need standup bases for
two Aaastra 480i phones. My provider cant get them, ebay doesn't have
them, Aastra is no help.
Anyone? I really do not want to have to pay top
Hi, all
Can someone give me an example on how to do following:
Asterisk receives incoming call from SIP
Asterisk asks for a pin number
Astersisk provides dialtone
Asterisk collects digits from the caller and places a call on another interface
Any pointers are greatly appreciated.
Thanks,
The DISA application should do what you are looking for.
PaulH
Rudolf Ladyzhenskii wrote:
Hi, all
Can someone give me an example on how to do following:
Asterisk receives incoming call from SIP
Asterisk asks for a pin number
Astersisk provides dialtone
Asterisk collects digits from the
On Monday 08 September 2008 23:00:50 Steve Edwards wrote:
On Mon, 8 Sep 2008, Tilghman Lesher wrote:
On Monday 08 September 2008 12:43:53 Nguyen wrote:
Hi listers,
We want to implement one call center with asterisk. The idea is it
should be scalable, with openser as an dispatcher and
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