Hello,
Is there any way of removing this line from showing on the console? I have a
script that logs in every few seconds to manager and it makes the CLI output
very hard to follow because of the == Parsing
'/etc/asterisk/manager.conf': Found. (Yes, Found! manager.conf was there 3
seconds ago,
Adrian A wrote:
Is there any way of removing this line from showing on the console? I
have a script that logs in every few seconds to manager (...)
Maybe a better solution is to rethink your architecture. The Manager API
is well suited for long running connections, so there is no need to
I installed festival following this guide (method 1)
http://www.voip-info.org/wiki/view/Asterisk+festival+installation
When I use english voices Festival works, when I change in Italian
voices, Festival return an error (generic).
Someone has faced and solved the same problem?
Thanks in advance
Hi All;
I would like to know if any one can helps or have idea
on how to disconnect the call in case no media (all
parties hanged up, but the channel still open) for the
following cases:
1) If we have IAX trunk, then how to disconnect the
call automatically after specific time (time out
media)
HI all,
I have set up a queue with 2 members (A B). 1st call is waiting
in the queue and a queue member A is ringing but don't take the call.
Member A keeps ringing. Then 2nd call is also get into the queue but
I found that queue member B doesn't ring. That's mean member B is
available to
On 10/04/2008, Stefan Reuter [EMAIL PROTECTED] wrote:
Adrian A wrote:
Is there any way of removing this line from showing on the console? I
have a script that logs in every few seconds to manager (...)
Maybe a better solution is to rethink your architecture. The Manager API
is well
On Fri, 28 Mar 2008 12:09:58 -0500, Peder @ NetworkOblivion wrote:
Any idea? If I remove call-limit on the sip.conf entries, it all goes
back to working fine. I tried 2, 9 and 99 on the call-limit and they
all have the same issues. I can't imagine why call-limit causes hints
to stop
On Wed, Apr 09, 2008 at 11:55:13PM -0700, Adrian A wrote:
Hello,
Is there any way of removing this line from showing on the console? I have a
script that logs in every few seconds to manager and it makes the CLI output
very hard to follow because of the == Parsing
Hello,
I've configured call Call forwarding using followme.conf. It works fine.
If no number pick up a call, I want to send the call to the voicemail.
How can I do this?
followme.conf
[150]
music=default
context=hledej
number=12,10
number=13,10
number=14,10
extension.conf
[from-internal]
Hello,
I want to get voice broadcasting system which have DTMF tone recognition.
I have no time to learn and istall it.
Is there anybody to sell me that system ?
Thanks,
Esref ALBAYRAK
___
-- Bandwidth and Colocation Provided by
Rilawich Ango Thursday, April 10, 2008 3:28 AM
I have set up a queue with 2 members (A B). 1st call is waiting
in the queue and a queue member A is ringing but don't take the call.
Member A keeps ringing. Then 2nd call is also get into the queue but
I found that queue member B doesn't
Hello
I have a couple of questions about running 1.4.17 on FreeBSD 6.3:
1 .On a FreeBSD host, In modules.conf, I naively removed the following
modules that I thought I didn't need, but after stopping/restarting
Asterisk, Zaptel stops reporting calls:
/usr/local/etc/asterisk/modules.conf
noload
On Thu, Apr 10, 2008 at 03:09:18PM +0200, Vincent wrote:
Hello
I have a couple of questions about running 1.4.17 on FreeBSD 6.3:
1 .On a FreeBSD host, In modules.conf, I naively removed the following
modules that I thought I didn't need, but after stopping/restarting
Asterisk, Zaptel
On Thu, 10 Apr 2008 16:14:01 +0300, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
chan_zap.so failed to load as it depends on res_smdi.so ?
I have no idea. Is there an up-to-date list somewhere, or some script
that lists dependencies for each module, so that we have some way of
knowing what can be
On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
Have you tried the using the SIPDtmfMode function in your dial plan?
Not sure how I would introduce that with my enum macro, but as a test I
did try it for this particular peer:
-- Executing [EMAIL PROTECTED]:1]
I'm having a major problem at one of my branch offices with Phantom
Rings on their asterisk-based phone system. The system was originally
built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC
card. The upgrade severely increased the frequency of the phantom
rings. I've
Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went to
background and connecting using -r, and colors disappeared for me as well. I'm
using screen as well (ls -l --color=auto works fine in screen too).
Is there a documented fix available, or is this more just an odd
Quoting Brent Davidson [EMAIL PROTECTED]:
I'm having a major problem at one of my branch offices with Phantom
Rings on their asterisk-based phone system. The system was originally
built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC
card. The upgrade severely increased
Brian J. Murrell wrote:
On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
Does anyone know if Asterisk will convert an inband DTMF from one sip
channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP
channel?
You might also try canreinvite=no for both your phone and the
On Wednesday 09 April 2008 17:18:32 Bob Pierce wrote:
We are using Asterisk 1.2.18 at this site. One of the users brought this
to my attention today.
We have a problem when we take the message off the voice mail. If I am
taking off the messages it used to be [on the old phone system] that no
Jon Pounder wrote:
I had the phantom rings for years, once a day same time roughly every
day, finally just got annoyed enough one day I trapped the telco on
the phone with me till I finally got to talk to the right person. The
right person knew instantly what I was talking about after
On Thu, Apr 10, 2008 at 11:43:18AM -0400, Joshua Kinard wrote:
Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went
to background and connecting using -r, and colors disappeared for me as well.
I'm using screen as well (ls -l --color=auto works fine in screen too).
Joshua Kinard wrote:
Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went
to background and connecting using -r, and colors disappeared for me as well.
I'm using screen as well (ls -l --color=auto works fine in screen too).
The colors work if you use the supplied
Quoting Brent Davidson [EMAIL PROTECTED]:
Jon Pounder wrote:
I had the phantom rings for years, once a day same time roughly every
day, finally just got annoyed enough one day I trapped the telco on
the phone with me till I finally got to talk to the right person. The
right person knew
Hmm, interesting, initially it wasn't working. Maybe I started it from outside
of screen? Odd.
BTW, is it possible for the SuSE script to support a variable to pass args to
the daemon? Like perhaps modifying ASTARGS to be a changable param at the top
of the script? I like the verbose
Hi,
I would like to be able to log call details in Asterisk. The kind of logs
that I like to generate is like this:
From
To Forward Time
Incoming Call604-343-3334
503-233-4454
Joshua Kinard wrote:
send core set verbose 999 when connecting in.
I'm running Mandriva and found the line that had -vvv. I like mine at
15, so just put 15 v's on that line. Worked great.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
On Thu, Apr 10, 2008 at 12:48:05PM -0400, Joshua Kinard wrote:
Hmm, interesting, initially it wasn't working. Maybe I started it from
outside of screen? Odd.
Or maybe it's plain buggy. Bug reports are welcomed.
BTW, is it possible for the SuSE script to support a variable to pass
Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I
edit /etc/asterisk/voicemail.conf with envelope=yes and after that I
left a message in a given mailbox near 11:00 AM. When a dial the
voicemail number in order to hear the message, the Astreisk server close
the cal and I get this
On Thu, Apr 10, 2008 at 01:08:21PM -0400, Doug Lytle wrote:
Joshua Kinard wrote:
send core set verbose 999 when connecting in.
I'm running Mandriva and found the line that had -vvv. I like mine at
15, so just put 15 v's on that line. Worked great.
15? What do you need that for?
On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote:
You might also try canreinvite=no for both your phone and the sip
peer.
Yeah, there is definitely no re-inviting going on. Both Asterisk and
the local handset are in a local network behind NAT with reference to
the SIP server that
(The only issue we've had with the TE121 is echo on voice calls, even
with the hardware echo cancelling module and lots of zapata.conf tuning...
did You EVER get the echo resolved ? How ?
We managed to get it tuned to the point where user complaints are
minimal, but there is definitely
On 10:00, Thu 10 Apr 08, Pete Kay wrote:
Hi,
I would like to be able to log call details in Asterisk. The kind of logs
that I like to generate is like this:
From
To Forward Time
Incoming Call
Tzafrir Cohen wrote:
On Thu, Apr 10, 2008 at 01:08:21PM -0400, Doug Lytle wrote:
Joshua Kinard wrote:
send core set verbose 999 when connecting in.
I'm running Mandriva and found the line that had -vvv. I like mine at
15, so just put 15 v's on that line. Worked great.
-Original Message-
Sent: Thursday, April 10, 2008 1:08 PM
Or maybe it's plain buggy. Bug reports are welcomed.
Nah, I think it was PEBKAC and PICNIC here :: sheepish grin ::
I tried adding --style options to the DAEMON var assignment, and it looks
like the -f check further down
-Original Message-
Sent: Thursday, April 10, 2008 1:22 PM
15? What do you need that for?
IIRC the highest verbosity level is 5. anything more than that doesn't
change the clogging of your logs.
Ah, 5 is max? Kinda like gcc not supporting anything greater than -O3? Good
to know
Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I
edit /etc/asterisk/voicemail.conf with envelope=yes and after that I
left a message in a given mailbox near 11:00 AM. When a dial the
voicemail number in order to hear the message, the Astreisk
On Thu, Apr 10, 2008 at 01:37:34PM -0400, Joshua Kinard wrote:
-Original Message-
Sent: Thursday, April 10, 2008 1:22 PM
15? What do you need that for?
IIRC the highest verbosity level is 5. anything more than that doesn't
change the clogging of your logs.
Ah, 5 is max?
Hi everyone,
I'm currently trying to enable call transfer to different domains in
asterisk box (Asterisk 1.2.13 running on Debian etch). I have a
configuration that requires me to transfer call to separate domains
like [EMAIL PROTECTED]:5050. My calls come from a R2 channels in a
board installed
On Thursday 10 April 2008 12:14:49 Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I
edit /etc/asterisk/voicemail.conf with envelope=yes and after that I
left a message in a given mailbox near 11:00 AM. When a dial the
voicemail number in order
On Thu, 2008-04-10 at 11:25 -0500, Tilghman Lesher wrote:
If you instead use a separate extension, you can use groups to
restrict the number of people accessing a particular mailbox:
Thanks Tilghman,
I didn't think of that. I'm sure that will work just fine for what we
need.
Have a great
Brian J. Murrell wrote:
On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote:
You might also try canreinvite=no for both your phone and the sip
peer.
Yeah, there is definitely no re-inviting going on. Both Asterisk and
the local handset are in a local network behind NAT with
I have an operator queue that is supposed to ring 2 phones, extension 10
and 11. Everything is working correctly but I keep seeing these
messages in my log: The device state of this queue member, Sip/10, is
still 'Not in Use'. Everything I've been able to find on this message
so far points
So this is just a general question, Is Asterisk really good?
Reliability?
Functionality?
Customization's?
I am coming from a Nortel world, were you pay for everything, and you
can't delve into the software. But it seems that customization would be
a great thing.
Like, setting up a
Hi Eugene,
Yes it's that good.
All the functionality you posted is possible.
Regarding your international calls, nothing more is required than two
asterisk servers (one at each location) and a broadband connection -
cards are only used to connect into pstn or isdn.
Why don't you start with a
On Thu, 10 Apr 2008, Eugen Soare wrote:
1 - Can you really make free outgoing calls from let's say Portland OR,
to Frankfurt Germany?
No. There is no free lunch. It takes electricity, bandwidth, and depending
on who you want to call in Germany, termination.
Thanks in advance,
On Thu, 2008-04-10 at 13:36 -0500, Brent Davidson wrote:
One more tidbit I just ran across in the upgrade.txt file, since you
mention NAT: In 1.4, you need to set canreinvite=nonat to disable
re-invites when NAT=yes. This is propably what you want. The settings
are now: yes, no, nonat,
Anything more than 'core set verbose 1' produces this message, however
verbose 1 does not display much of anything.
On Thu, Apr 10, 2008 at 1:53 AM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Wed, Apr 09, 2008 at 11:55:13PM -0700, Adrian A wrote:
Hello,
Is there any way of removing this
On Thursday 10 April 2008 02:14:17 pm Steve Edwards wrote:
1 - Can you really make free outgoing calls from let's say Portland OR,
to Frankfurt Germany?
No. There is no free lunch. It takes electricity, bandwidth, and depending
on who you want to call in Germany, termination.
though you
There is an OpenSER proxy in front of Asterisk which handles the clients.
The script is called by OpenSER whenever a client sends a SUBSCRIBE request
for MWI. It uses php to connect to Asterisk like so:
fsockopen($mhost,5038, $errno, $errstr, 5) and gets the user's voicemail
counts.
I'm not sure
Steve Edwards wrote:
On Thu, 10 Apr 2008, Eugen Soare wrote:
1 - Can you really make free outgoing calls from let's say Portland OR,
to Frankfurt Germany?
No. There is no free lunch. It takes electricity, bandwidth, and depending
on who you want to call in Germany, termination.
Yep.
BTW Eugene, was just reading Tom Keatings blog (he's a pretty well known
reporter around here), he was just talking about a new commercial
appliance called CogoBlue that you could send to the person at the other
end if they don't know anything about voip or pabx's to make it even
easier for them
Remember - True TELCO grade systems simply cannot be compared to
anything else.
You want the reliability , uptime, and all the bells and whistles of
true Carrier Grade Hardware/Software
then you pay for it. If you want something you can tinker under the
hood with and do some cool
stuff with,
Brent Davidson wrote:
I have an operator queue that is supposed to ring 2 phones, extension 10
and 11. Everything is working correctly but I keep seeing these
messages in my log: The device state of this queue member, Sip/10, is
still 'Not in Use'. Everything I've been able to find on
Tzafrir Cohen wrote:
On Thu, Apr 10, 2008 at 01:08:21PM -0400, Doug Lytle wrote:
Joshua Kinard wrote:
send core set verbose 999 when connecting in.
I'm running Mandriva and found the line that had -vvv. I like mine at
15, so just put 15 v's on that line. Worked great.
Asterisk as a PBX is fantastic. It offers the features found in the most
sophisticated traditional
Until now the third party PBX configuration software has not had the
sophistication of
Asterisk product itself. Check out cogoblue.com It is a visual drag and
drop configuration tool that
is
Just a thought. A while back there was discussion about the merits of
having a product (in that case an O/S) with contracted vendor support
or relying solely on list support.
I note in the post below where one responder states
It may also have been because less than 23 hours had elapsed
I installed the Digium T1 card on Dell Poweredge 2950 and the system crashed
several times, we got a Kernel Panic and first though it was the OS so I
switched from Fedora 7 to Centos 5.1.
Our server was alarming in our monitoring system, when our Infrastructure
department investigated the issue
On Thursday 10 April 2008 15:43:33 John Signorello wrote:
Asterisk as a PBX is fantastic. It offers the features found in the most
sophisticated traditional
Until now the third party PBX configuration software has not had the
sophistication of
Asterisk product itself. Check out cogoblue.com
Any time you have this kind of hard lockup with a Digium card you should
run, not walk to the nearest phone and call them.
broadband Voice wrote:
I installed the Digium T1 card on Dell Poweredge 2950 and the system crashed
several times, we got a Kernel Panic and first though it was the OS so
On Thu, Apr 10, 2008 at 04:05:47PM -0400, Al Baker wrote:
Remember - True TELCO grade systems simply cannot be compared to
anything else. You want the reliability , uptime, and all the bells
and whistles of true Carrier Grade Hardware/Software then you pay for
it. If you want something you can
Brent Davidson wrote:
I'm having a major problem at one of my branch offices with Phantom
Rings on their asterisk-based phone system. The system was originally
built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC
card. The upgrade severely increased the frequency of the
On Thu, 2008-04-10 at 16:05 -0400, Al Baker wrote:
Remember - True TELCO grade systems simply cannot be compared to
anything else.
You want the reliability , uptime, and all the bells and whistles of
true Carrier Grade Hardware/Software
then you pay for it. If you want something you can
BUG: soft lockup detected on CPU#1!
[c044b2a4] softlockup_tick+0x96/0xa4
[c042e214] update_process_times+0x39/0x5c
[c04196ff] smp_apic_timer_interrupt+0x5b/0x6c
[c04059bf] apic_timer_interrupt+0x1f/0x24
.
You don't happen to be running a XEN Kernel are you? I
Quote
And for what it's worth, at my new job I babysit about a 16-machine
cluster running VICIdial for close to 200 agents, and by and large, it
just runs. It's got about 20 T-1s feeding it
Please share more about this.
What/How are you clustering the boxes ?
Is this all VOIP or TDMF front
We're using PAE Kernel.
On Thu, Apr 10, 2008 at 4:30 PM, Michael L. Young [EMAIL PROTECTED] wrote:
BUG: soft lockup detected on CPU#1!
[c044b2a4] softlockup_tick+0x96/0xa4
[c042e214] update_process_times+0x39/0x5c
[c04196ff] smp_apic_timer_interrupt+0x5b/0x6c
[c04059bf]
http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.html
I thought others would find this helpful.
I have no tie to any vendor or company referenced in this paper and I
believe the same concepts to be applicable to the Asterisk world.
Lee Jenkins wrote:
Brent,
I had a similar problem and I feel for you, its frustrating.
Are you using polycom phones by chance? Here is the problem that I had, not
sure if your problem is related.
Specs:
- 6 Polycom 301 phones.
- CentOS 4 Server with Asterisk 1.2.x
- Sangoma A200 card
On Thu, Apr 10, 2008 at 12:21:46PM -0700, Adrian A wrote:
Anything more than 'core set verbose 1' produces this message, however
verbose 1 does not display much of anything.
So obviously there are two ways to resolve thins (apart from leaving
things as they are now) - demoting the priority of
I saw an add-on for Asterisk at
http://www.yosd.at/index.php?option=com_contenttask=viewid=23Itemid=38
Anyone tried any of these ?
How well did they work ??
Asterisk Applications Print
http://www.yosd.at/index2.php?option=com_contenttask=viewid=23pop=1page=0Itemid=38
The Asterisk PBX
Hello,
It might not be Digium's fault, I ran into similar problems with Dell
2950 servers and other PCIexpress cards. I even went so far as to have
several components replaced by Dell on one of the affected servers to
no avail. After many months of banging my head against a wall I
stumbled across
Thanks. I have checked that the queue.conf. I keep the default
setting as autofill=yes in my tests. That's mean even autofill=yes,
the 1st caller will still stick the whole queue.
asterisk version : 1.4.18
--queue.conf--
; AutoFill Behavior
;The old/current behavior of the queue has a
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o
LD [M]
On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote:
Please share more about this.
What/How are you clustering the boxes ?
Is this all VOIP or TDMF front and VOIP for agents in back ?
What kind of Boxes ? What O/S
What tools are you using to monitor this big-azz mother ?
Rilawich Ango wrote:
Thanks. I have checked that the queue.conf. I keep the default
setting as autofill=yes in my tests. That's mean even autofill=yes,
the 1st caller will still stick the whole queue.
asterisk version : 1.4.18
--queue.conf--
; AutoFill Behavior
;The old/current
I am exploring the contacts directory in Polycom and I am wondering if a
big -directory.xml on the boot server will eat up the memory
and crash the Polycom phone once downloaded onto the phone.
The asterisk directory extension is good but because users cannot see
the names I thought to
Hi
I am wondering if anyone has experince running multiple instances of
Asterisk using Debian vserver. The senario I want to implement is to have a
couples of DIDs. Some DIDs are handled by Asterisk instance #1 and some
DIDs are handled by Asterisk instance #2. The two Asterisk are within
Hi Guys,
I have a TDM400P (2FXS and 2FXO) installed on my asterisk (1.4.18.1)
Zaptel (1.4.9.2) running on Fedora Core 7, below are my configurations
and some diagnosis I did. The problem is when I connect an analogue
phone on either of the FXS channels I don't get a dialtone, I can't
call any of
Hi,
I'm not looking for a programma that show the queue logging.
But is there a way to check during a call, which member is connected to
the caller.
Kind Regard,
Arjan Kroon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel
1.4.10. They have the hardware echo cancellers. I am having an issue
though, when i talk, it cuts out the other end. So for example, i
called up another asterisk box and was listening to the prompts and as
they were playing if
Has anyone gotten Realtime and Macros to work with Asterisk? Or is it
better to run the macros directly from the flat conf file?
I have the macro working without using ARA but as soon as I use ARA it
fails.
Any assistance or pointing me in the right direction is appreciated.
Hi All,
I am having problems with some SIP peers. I seem to loose registration.
If I reload SIP the registration comes back. They usually stay
registered for about 2 days before they drop. The problem is not all of
them drop usually just the list 2 in the list. The other strange thing
is
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