On Thursday 09 February 2006 11:28, Kevin P. Fleming wrote:
There are examples (IIRC) of making the phone auto-answer for specific
types of calls; those should get you started, since they demonstrate how
to have the phone choose a different 'alerting' configuration on a
call-by-call basis.
Here is another log from the * server CLI, I reall hope some one can help me out on this one. thanks
|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and
campaign_id = '' and call_time and lead_id != '';|-- VDAD get agent: |0|update of vla table:
I have running * without any Digium (or any other) hardware. Now I need to
connect analog FAX machine to it. I think that cheapest and easiest way is
to buy ATA. Please correct me if I'm wrong.
Since you have no Digium hardware (and thus no connection to POTS or
PRI)... are you routing your
How come I don't have the MeetMe application registered?
Regards,
Anthony.
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Hello list,
I am currently doing a job for a summer camp. They
would like to have several phones around the camp from
which people can call in to the main office. It is an
older campus and it is comprised of mostly old
nungalow type housing. I need to install these phones
several hundred yards
I have some more info about my deadlocks...
It usually happens when you have a callwaiting and users are pressing flash
button
on ZAP channels.
Here is what I see on CLI
Feb 9 14:48:41 WARNING[4225]: channel.c:787 channel_find_locked: Avoided
deadlock for '0x9655d18', 10 retries!
Anyone
On Thu, Feb 09, 2006 at 10:39:43AM -0600, Miguel wrote:
Tzafrir Cohen wrote:
Makefile:204: target `ztdummy.o' given more than once in the same rule.
Something is bad. Did you edit Makefile?
Yes. I delete all the modules, leaving the lines like this
before:
At 10:29 PM 02/08/2006, you wrote:
That's what I figured; I was really wondering whether AMP would
specifically *prevent* that kind of configuration.
I have a similar configuration, 4 lines, 4 companies and started with
AMP. While I can't tell you it's not possible, I can tell you I
looked
Dov Bigio wrote:
Is there a way to rotate CDR CSV files via Asterisk, or should I handle this
outside Asterisk?
As another reply already mentioned, no and yes :-)
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As of Asterisk 1.2, a Master.csv file is also created in the
cdr-custom directory by default. I had a horrible time until I figured
that out. Any reason this was done by default that two Master.csv
files are created(cdr-csv/ and cdr-custom/)?
MATT---
On 2/9/06, Dov Bigio [EMAIL PROTECTED] wrote:
You should really post VICIDIAL questions on the
[EMAIL PROTECTED] list instead of the general
Asterisk-users list.
It looks like you don't have the 'o' flag on your Dial exten in
extensions.conf. The callerID is not being preserved so there's no way
to follow the call.
MATT---
On 2/9/06, Vic
Could anyone either recommend a website or howto on optimizing Linux to
run asterisk. Such examples of what I mean are..
Renice of asterisk pid's
Forcing irq smp_affinity (For interupt hogging T1 cards)
.. That kind of stuff, I looked on the wiki and nothing directly
mentions server
Thanks so much to all of you this has helped me out immensely !
Have a great day !
Nora
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Sent: Thursday, February 09, 2006 7:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
BOFH told me he uses it to listen to his co-workers
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, February 09, 2006 12:27 PM
To: asterisk-users@lists.digium.com
Subject: SOLVED: Re: [Asterisk-Users]
This feature also works on the IP301 phones. The obvious caveat is that
it is one-way only. Still nice for an all-page though.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, February 09,
Title: Manager API 'Redirect' is not working for both end of a call.
Hi everyone,
Here's what I come across. Phone A calls phone B through Asterisk and they are talking. An application uses manager api caught the channel IDs of both legs of the call. It then issue 'Redirect' with both
Title: Connecting to live calls
Thnx.
Checked them out. There still a lot needed to be desired.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Alexander
LopezSent: Wednesday, February 08, 2006 11:24 PMTo:
Asterisk Users Mailing List -
Florian,
What exactly do you mean by seperating traffic in to differt SIP peers?
The situation is as follows:
I have OpenSer connected to our SIP provider/PSTN Provider (the answer
to your question: Enertel).
Asterisk registers to OpenSer, which then forwards the call to PSTN.
Asterisk
hi ive been tryin to get oh323 to work and installed it without any problems
but it gives me the same error all the time this is the third time ive
installed it..please if anyone can kindly help me out thanks in advance...
[chan_oh323.so]Feb 10 00:35:29 WARNING[4891]: loader.c:258
On Wed, 2006-02-01 at 10:07 -0200, Juan Carlos Castro y Castro wrote:
Juan Carlos Castro y Castro wrote:
I would not recommend running that much load in a single server of FXS,
because the power supply requirements are too great, but if at least
half is FXO then you should be fine.
Hey all,
I'm having problems where there is significant static when making SIP -
PSTN calls. SIP - SIP and SIP - VM calls are totally clear and fine.
Here's the setup:
Polycom 601,501, and ten 301s.
Digum 2400 TDM card w/echo cancelling, 12 FXO ports.
The TDM card is on IRQ 5 with nothing
Could we have run into another Americanism here?
OK, back to being English and bashing the French ;-}
Nice french 'tache :)
Anyway, the problem at hand is not the french language here. The problem
is that any worthwhile i18n mechanism must pass the yota master test.
Speak like this, it
On Feb 9, 2006, at 10:50 AM, Gerard Saraber wrote:
Aha! good to know, I am running asterisk 1.2.4 and zaptel-1.2.3, should
I switch to CVS ? I've tried the MG2 canceler with the above versions,
each time I tried it, I had a constant echo, where with the mark3 it
went away after a second or
Jean-Michel Hiver wrote:
Hi List,
Do you know if there are any plans to improve i18n for Asterisk? The
current i18n way of doing it with asterisk is very limited and most of
the time does not work.
For example, take voicemail:
message received at seven 30 am might sound good in English.
Sam Tam wrote:
I think this is a question that has been discussed before.
But you see nowadays most carriers will provide thing like SIP using IP
authorization rather than username and password and I am now wondering
whether Asterisk can do something like that or not?
In the voip channels
Greetings to All,
I hope someone has already gotten this working. I spent all day today trying
to get ooh323 and gnugk to run on the same box. After a lot of tweaking to
get everything compiled, I got both up and running.
I can make calls IAX to H323, but cannot make calls in the reverse
Keep a watch here as I believe there is a bug brewing...http://bugs.digium.com/view.php?id=6147cheers,MOn 10/02/06,
Bartosz Jozwiak [EMAIL PROTECTED] wrote:
I have some more info about my deadlocks...It usually happens when you have a callwaiting and users are pressing flashbuttonon ZAP
I recently upgraded to 1.2.x (not quite sure which version, whatever it is I downloaded it on 1/30/06.)
At the time of the upgrade, I had a Nufone toll-free number going into
my IVR. Extension 2000 rang the Cisco phone on my desk, and the
caller id came through just fine.
As soon as the
anyone else having issues with voipjet? i am getting nothing but dead air and a hanging iax channel.
-yair
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Yes, it's not working for me either. First time it's gone down for me.--AndrewOn 2/9/06, Yair Hakak [EMAIL PROTECTED]
wrote:anyone else having issues with voipjet? i am getting nothing but dead air and a hanging iax channel.
-yair
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On Thu, 2006-02-09 at 14:16 -0600, Matthew Fredrickson wrote:
Try MG2 with trunk and KB1 with 1.2. KB1 is supposed to be fairly
reliable in 1.2, and MG2 in trunk has a good possibility of
outperforming KB1 from 1.2.
Matthew Fredrickson
Thanks! testing it now, on my test calls it appears
I doubt a system would handle two, aside from the power draw the heat is radical.RobOn 2/9/06, Hans Witvliet
[EMAIL PROTECTED] wrote:On Wed, 2006-02-01 at 10:07 -0200, Juan Carlos Castro y Castro wrote:
Juan Carlos Castro y Castro wrote: I would not recommend running that much load in a
Same here. The call is accepted,
but never makes any progress. I tried all four servers with the same results.
Wes Baehr
Ability Business Computing, Ltd.
Office: (330) 882-0455 x25
Cell: (330) 990-9445
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi Noah,
You've run into the same problem a lot of other people have had. Remapping
hard keys works fine, but remapping soft keys does not. In fact, trying to
remap the soft keys results in some pretty weird behavior. The Polycom
manual is a little misleading in that it doesn't mention this at
Hi -
Has anybody been able to successfully remap SpeedDials on Polycom phones?
The manual seems to indicate that you can, and I followed the advice in this
list message:
http://lists.digium.com/pipermail/asterisk-users/2005-October/129142.html
The result I get is that the remapped buttons act
For prepaid billing I would use astcc. It comes with
asterisk (it may be in an extra download but is made
by digium to work with asterisk). For post billing
there are a lot of diffrent solutions out there. Do a
google search. As far as putting in the DID info you
would put it in to your dial plan.
Same thing here. I had this problem awhile ago and made this
workaround.
Going to another trunk does not work because they are answering and not
sending a error code. If you are using AAH code then this waits 10
seconds on your Voip then times out and goes to PSTN. You can modify
for your
FYi,
A zttest reports:
--- Results after 144 passes ---
Best: 100.00 -- Worst: 99.890137 -- Average: 99.993462
Roman
Quoting Roman Volf [EMAIL PROTECTED]:
Hey all,
I'm having problems where there is significant static when making SIP
- PSTN calls. SIP - SIP and SIP - VM calls are
I think your problem is the Dell 650. What are the
specs on it ? If you want a system that can support
200 users you will need to do a lot better than that.
Also you will be dealing with T1's/E1's and not POTS
lines. I think a good place to start (if you havent
already) is the book that has come
Hello All,
I'm looking to get some feedback on which solution of providing FXS is
going to have the best results with faxing. I'm only looking to see what
method is going to provide the best digitization into Asterisk, not for
transmission from Asterisk to else where. Any recommendations of
Hi,
According to what I read, for g.729A 1 line I need 21 kbps.
Now is it 21 coming in and 21 going out? Or 10.5 coming in
and 10.5 going out?
Assuming that it is 21 coming in my traffic would be
21x3600=75600kb = 9450KB /per hour So close to 10M of traffic
is coming into my network per
Hi Dovid,
Thank you for the book. I'm already reading it.
I have a dell 650 server, 1Gig of memory, 1 CPU (3.07Ghz). What
hardware would you recommend for the 200 users w/ about 20 concurrent
calls ?
As always I thank you so much for your help.
Nora Lavelle
-Original Message-
Hi Ronald,
Ronald Voermans wrote:
What exactly do you mean by seperating traffic in to differt SIP peers?
The situation is as follows:
I have OpenSer connected to our SIP provider/PSTN Provider (the answer
to your question: Enertel).
Ah 'kay.
Asterisk registers to OpenSer, which then
Hi all, please forgive me relative lack of knowledge with
Asterisk, but Ive not played with PBX systems for a while and Im
just re-finding my feet.
Ive set up my first Asterisk server, I have it
configured with a Digium X100P Analogue pots board, I have my Called ID working
and everything
Hey All,
Ive been working on trying to get asterisk to play
nice under Xen and Ive run into a bit of a road block.
Im not using any hardware stuff only ztdummy.
First I had issues getting ztdummy to work but that was
solved by recompiling the xenU kernel to have CONFIG_CRC_CCITT=y
Anthony Azzopardi wrote:
How come I don't have the MeetMe application registered?
You need a timing source. See:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
Kevin
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Have you tried System?
On 2/9/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi all, please forgive me relative lack of knowledge with Asterisk, but I've
not played with PBX systems for a while and I'm just re-finding my feet.
I've set up my first Asterisk server, I have it configured with
Specifically:
exten = s,1,System(/usr/sbin/myperlscript.pl ${CALLERIDNUM})
will execute myperlscript.pl with the caller id as an argument as the first
priority.
hth
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 09, 2006 4:00 PM
To: Asterisk Users
Seri,
I think you might just need to find the
right variable and pass it to your script. Also, youll probably
need to read up on AGI Im assuming thats how you would
launch your Perl scripts. Heres the list of variables on the wiki:
http://www.voip-info.org/wiki-Asterisk+variables
Or you could just use System
and do it the easy way! AGI is better suited for more complicated scripting. In
any case, check out the TFOT book youll like it.
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Thursday, February 09, 2006
On Wed, 2006-02-08 at 08:35 +0100, stoffell wrote:
On 2/6/06, Conrad Wood [EMAIL PROTECTED] wrote:
Please note the spelling of uniqueid. I find the spelling in
res_features.c - but only once I patched it with bristuff patches.
Does anyone know whether that is a known problem with bristuff?
Get to answer my own post. I found an article that talks about the need for
1000HZ timing in the kernel for ztdummy to work properly. Xen's kernel
builds default to 100HZ just like 2.4 kernels.
I changed the values to 1000 in
/xen-3.0.0/linux-2.6.12-xenU/include/asm-xen/asm/param.h and
Thanks for the replies, the System command was exactly what I needed and I
have now modified my calling plan and it looks like it's going to work
perfectly (it's a little late at home to try testing now in the UK).
My next job is going to be getting to grips with the various codec's and
working
Sorry to disagree, but unless you are transcoding or have a significant AGI IVR or something, that server should be just fine. While I prefer a bigger server for most installs, it is not because it is needed, more of a, money well spent kind of thing.
If you are transcoding and have an IVR, the
On Thu, Feb 09, 2006 at 06:23:26PM -0500, John Cianfarani wrote:
Get to answer my own post. I found an article that talks about the need for
1000HZ timing in the kernel for ztdummy to work properly. Xen's kernel
builds default to 100HZ just like 2.4 kernels.
I changed the values to 1000 in
P-3 Xeon 550 w/ 23 B-channels concurrent - 150 SIP
clients:
PID USER PR NI
VIRT RES SHR S %CPU %MEM
TIME+ COMMAND9712 asterisk 15 0 81556
11m 4392 S 18.9 0.4 11:19.06
asterisk
CPU is overrated.
-Original Message-From: Richard Amerman
[mailto:[EMAIL PROTECTED]Sent:
Hi!
I am new with asterisk and I have my first problem with the attended
call transfer feature.
When a call comes in, i take the call and i would like to transfer it.
So I press the * button (mapped for the attended transfer in
features.conf) and the number for the receiving extension.
The
Anyone know of one that I could use?
Thanks,
Shri
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Kevin P. Fleming wrote:
Dov Bigio wrote:
Any way, if any developers are reading this, I don't think that rotating
asterisk logs is the best way to handle this problem!
Maybe a more user-friendly message could be logged, infoming which file
reached the 2.0GB.
Unfortunately when we
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Thursday, February 09, 2006 6:33 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk and Xen
On Thu, Feb 09, 2006 at 06:23:26PM -0500, John Cianfarani wrote:
This is hopefully on topic. I'd like thoughts on this. I'm looking at
doing some dialplan work which would grab the sip devices IP number. If
that ip number is in an allowed list, the call would be allowed to go
through otherwise congestion would be passed. Any thoughts?
Darren Wiebe
It errors when you ask it for the channel 'test-1' because the parameter
is the channel name, not the peer name. I've used 'show channels
concise' and then parsed the output in the past.
Peter Hoppe wrote:
Hello!
I have an asterisk setup where several sip devices are connected to an
Hi all,
I had problem running MySQL on FC3 and what I found from googling was
that SELinux should be disabled to make MySQL work n FC3. Now I am
concerned about Asterisk, is it a good idea to disable SELinux. Or is
there any other way to make MySQL work without disabling SELinux?
Thanks,
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Voicemail
Problem
I have just setup my
OPENSER to work with the asterisk 1.2.2.
I've
Please help for this. I really got stuck at this. After
a few tries , asterisk refuses connection anymore until the previous connection
timeout.
Let me know if you require more
info.
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday,
this is a Normal behaviour, nevertheless i dont think is a correct
behaviour. Several weeks ago other user asked the same, i suggested him
to open a feature request on bugs.digium.com, check for that
regardsOn 2/9/06, Thomas Artner [EMAIL PROTECTED] wrote:
Hi!I am new with asterisk and I have my
Hey Tim,My 2800 does H.323 to a CCM and SIP to my asterisk box. I actually don't forward calls directly to my asterisk box from the 2800. As Juan pointed out, you need to set up your dial peers so that your 2600 knows what to do with the calls. I'm not a guru when it comes to configuring Cisco
Hi,
You need the unattended transfer (blind transfer) featuer. That implemented
in Asterisk (#) button. Not attended transfer.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Artner
Sent: Friday, February 10, 2006 8:01 AM
To:
You don't have 'vm-goodbye' voice file. Check under
/var/lib/asterisk/sounds
Wojtek
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users]
Yeah something like that sound to be right.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: Friday, February 10, 2006 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP
Can you be more detail about the setup?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E
Johansson
Sent: Friday, February 10, 2006 4:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/goodbye.gsm/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks for replying.
Are there any other ideas ?
Regards,Sam
From: [EMAIL PROTECTED]
Or perhaps slow them down or pipe to a file?
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After installing the timing source , what do I have to do to get meetme
application registered? Do I have to recompile asterisk again ? I don't
see the compiled meetme.so module in the directory.
Regards,
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On 2/10/06, Eric Bishop [EMAIL PROTECTED] wrote:
Or perhaps slow them down or pipe to a file?
I usually run Asterisk in a screen session, and use Ctrl-A, [ to
scroll through screen's buffer... I'm sure there's other ways too :)
Andrew
--
Linux supports the notion of a command line or a shell
Has anyone every attempted to set up a mini PC to achieve much the same
functionality as the fonebridge box?
The sort of thing I'm imagining is a micro itx board case in a
completely solid state configuration (flash disk, maybe a psu fan but
only if really required), with a TE210P (or equiv)
You are looking for vn-goodbye, most likely under
sounds/vm
W
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 9:21
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
It is also there ..
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/vm-goodbye.gsm/var/lib/asterisk/sounds/vm-goodbye.gsm[EMAIL PROTECTED]:/home/sam#
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 10:59
Sam Lee wrote:
After installing the timing source , what do I have to do to get meetme
application registered? Do I have to recompile asterisk again ? I don't
see the compiled meetme.so module in the directory.
Yes, you need to compile zaptel, and then recompile/install asterisk.
Kevin
I am having a strange problem with an asterisk servier using R2 Unicall
in Mexico. Most calls go through fine but some of them give me an error like
this:
-- Executing Dial(SIP/86-db41, Unicall/g2/014448343600) in new stack
-- Called g2/014448343600
Feb 9 21:44:39 WARNING[23069]:
Hi all,
Just a quick question, when typing show queues from the command line,
asterisk outputs queue stats to the screen.
I was wondering if anyone could tell me what these values are;
W:0, C:0, A:0, SL:0.0% within 0s
I'm guessing that SL is service level, C is probably completed calls or
Has anyone used a
Brooktrout T114 Card with Asterisk ?? If so where can i find the drivers for the
brooktrout Card
I am running
Asterisk 1.2.4 on Red Hat 9
Tertius
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You can use the following:
switch3*CLI show function SIPCHANINFO
switch3*CLI
-= Info about function 'SIPCHANINFO' =-
[Syntax]
SIPCHANINFO(item)
[Synopsis]
Gets the specified SIP parameter from the current channel
[Description]
Valid items are:
- peeripThe IP address of the
Hack hack hack 8-) Now - comments inline...
Here's the log of verbose level 3
Asterisk*CLI
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing
Yes,
But without going deeper into OpenSer (since this IS a Asterisk list):
With OpenSer I'm using RTPPRoxy. I don't think i can manage rtpproxy to
bind to multiple addresses. I'll look for that anyway.
Thanks,
Regards,
Ronald.
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
On Fri, Feb 10, 2006 at 01:22:31PM +1100, Eric Bishop wrote:
Or perhaps slow them down or pipe to a file?
Send them all to a log file as well. See logger.conf .
tail -f log.file | grep whatever
--
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http://tzafrir.org.il |
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