> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
> k. Creasy
> Sent: Friday, October 14, 2005 10:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] No Audio from Console but mpg123 from shell
> work
--- Jonathan Lin <[EMAIL PROTECTED]> wrote:
> you get ping time in the status page if your extension.conf has
> qualify=yes
Setup
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qualify=yes
sip show peers
Name/user Host
you get ping time in the status page if your extension.conf has qualify=yes
Quoting Samy Antoun <[EMAIL PROTECTED]>:
--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:
Hmm.. What is the output of "sip show users" and "sip show peers"?
sip show users
Username Def.Context ACL NAT
200
Walter Klomp wrote:
Hi,
After only 4 days I have 107472352 bytes in 46007 allocations in file
'pbx_spool.c'
I seem to recall a memory leak in pbx_spool being fixed a few days ago.
check the asterisk-cvs mailing list archive on lists.digium.com
__
Sorry, I was pretty busy and did not work on my *.
Problem was in zaptel not properly registering driver with udev.
Manually updating udev rules fixed the problem.
Thanks,
Rudolf
- Original Message -
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, October 11, 2005 6:20 A
Hi,
After only 4 days I have 107472352 bytes in 46007 allocations in file
'pbx_spool.c'
asterisk*CLI> show memory summary
180 bytes in 2 allocations in file 'netsock.c'
12 bytes in 1 allocations in file 'devicestate.c'
2268 bytes in 1 allocations in file 'jitterbu
I don't know how to make this happen, and I don't even think it is
really possible given the current Queue app, but this would be a very
nice feature to have. The queue shouldn't pass a call to an agent if
they are already on a call from the queue, but an incoming call from
another internal
Yes, is defined in voicemail.conf too. The context is the glue of the
system, this is what I understood, the way to follow, the arrow that
show the direction to every application.
But the problems, still remain.
2005/10/16, Jason Walker <[EMAIL PROTECTED]>:
> Correct - but is the context defined
Setting incominglimit = 1 does not really solve the problem as I had
already mentioned. That practically takes away the call waiting and will
block all incoming calls including direct dialed calls. She does not
want that. Moreover, incominglimit is deprecated too.
-- jt
On Sat, 2005-10-15 at 22:
Have you tried the "incominglimit" parameter (or did she)?
I have found this to work pretty well when limiting the number of calls.
After monitoring the "full" log, I saw that incoming calls where
incrementing or decrementing the active call parameter for SIP agents. By
limiting the number of call
Correct - but is the context defined in voicemail.conf? As mickeymouse? Or
whatever...?
;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Saturday, October 15, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subje
I have compiled the OH323 module for my system.
When can I find some info on how to properly configure it?
I haven't read any info for its configuration, and I need some starting info.
Were do I start?
Obelix
This message was s
One of my friends is facing this problems and I could not find any
solution to that. Hence this post.
In her Asterisk PBX, she has programmed about 10 agents, and strategy is
rrmemory. Everything works fine. When an agent has received an ACD call,
another call is not presented to him as long as he
well, is just the context.
You could call it as you prefer, mickeymouse???
;o)
Bye
2005/10/16, Linc Fessenden <[EMAIL PROTECTED]>:
> FaberK wrote:
> > [EMAIL PROTECTED]
> > ---
> > Some ideas?
>
> Only thing I have that even looks different is
> [EMAIL PROTECTED]
>
> --
> -Linc Fessenden
>
> In th
First answer:
the envelope came back to me!
Yesterday I've added the line:
notifymimetype=text/plain
into sip.conf, so no more envelope just text into logs.
Is just an answer to you, Linc.
Nothing for me, yet.
2005/10/16, FaberK <[EMAIL PROTECTED]>:
> I've got it too until yesterday!!!
> Now no m
FaberK wrote:
[EMAIL PROTECTED]
---
Some ideas?
Only thing I have that even looks different is
[EMAIL PROTECTED]
--
-Linc Fessenden
In the Beginning there was nothing, which exploded - Yeah right...
___
--Bandwidth and Colocation sponsored by Easyn
Anyone have anything on this? (I'm sure someone will complain about me
bringing it up again, chill out...)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, October 14, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Comm
What is the adit 600 doing? FXO? FXS? how you connected to the PSTN?
I got an Adit 600 with both FXO and FXS as well as a PRI and I'm
getting CallerID on all three.
On 10/14/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hello All,
> Just a question, I have an adit600 and I am looking for a wa
I've got it too until yesterday!!!
Now no more envelope either.
This is from extensions.conf:
---
exten => 221,1,Dial(SIP/221,20,tr)
exten => 221,2,Voicemail(u${EXTEN})
exten => 221,102,Voicemail(b${EXTEN})
exten => 221,103,Hangup
---
this is from sip.conf:
---
[221]
type=friend
username=221
secret
FaberK wrote:
Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm ab
Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm able to send the r
The appendices are in the book. :)
Apparently only missing in the pdf.
> Is appendix A and B missing?
>
> > -Original Message-
> >
> > Leif Madsen a écrit :
> >
> > >Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
> > >Documentation Project, in conjunction with O'Rei
Maybe you don't have libc6-dev installed?
On 10/15/05, Brent August Torrenga <[EMAIL PROTECTED]> wrote:
Having trouble running make on res_perl:[EMAIL PROTECTED] res_perl]# makeperl -MExtUtils::Embed -e xsinitgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/-I/usr/src/asterisk-1.0.9
Having trouble running make on res_perl:
[EMAIL PROTECTED] res_perl]# make
perl -MExtUtils::Embed -e xsinit
gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/
-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asteri
Is appendix A and B missing?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jean-Michel Hiver
> Sent: Saturday, October 15, 2005 5:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Discussions regarding The Asterisk Docum
--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:
> Hmm.. What is the output of "sip show users" and "sip show peers"?
sip show users
Username Def.Context ACL NAT
200 from-internalNo No
210 from-internalNo Always
310 from-internalNo Always
sip show peers
Name/u
Leif Madsen a écrit :
Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.
In the tr
On Saturday 15 October 2005 16:18, [EMAIL PROTECTED] wrote:
> Well - number 1 - it IS - CVS HEAD.
Agreed.
> Next - I always run "make samples". In the /etc/asterisk/ directory it
> renames all your old config files that have changed to *.old.
>
> So long as you don't stop and restart Asterisk -
You can find RPMS with bristuff included at:
http://www.laimbock.com/asterisk/
the are compiled for centos.
rebuilding the SRPMS under FC3 work without a problem.
> We have a QuadBRI ISDN card from Digium. We would like to make it work
> with Fedora Core 4 (maybe FC3), but haven't succeeded. Comp
Hmm.. What is the output of "sip show users" and "sip show peers"?
On Sat, 2005-10-15 at 12:30 -0700, Samy Antoun wrote:
--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:
> Are the devices at 200 and 310 set up to register with your asterisk?
Yes, they are registered and I can call them
I´m a new user.I´m reading it seams to be very good.
Great job. It´s realy the book i as tryng to find!
Regards!
Cumprimentos,
André Rodrigues
Grupo Paulo Serra & Irmãos, Lda.
Direcção de Sistemas de Informação
Tel.: +351 25237 (ext: 296)
Fax: +351 252313483
Telem : +351 964245524
E-ma
Hello Leif , The appendices A & B are missing from the zip file
available at the location mentioned below . Is there some reason of
copyright that is not mentioned here ? Tia , JimL
On Sat, 15 Oct 2005, Leif Madsen wrote:
Jared Smith, Jim van Meggelen, and Leif Madsen
On 15 Oct 2005, at 19:58, Leif Madsen wrote:
Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in
On 10/15/05, Lars Dybdahl <[EMAIL PROTECTED]> wrote:
It seems that the bristuff source from junghanns.com wasn't written
for gcc 4, which is the one included in FC4, and I have seen somedescriptions on making zaphfc compile, but there are more problemsthan just that one. Also, RPMs would reduce the
On Sat, 2005-10-15 at 21:35 +0200, Lars Dybdahl wrote:
> I had a typo in my original mail - of course it's from junghanns.net,
> not from digium, and so is the bristuff, that I downloaded.
>
> I'm not searching for solutions on how to make it compile - I'm just
> trying to find out if anybody succ
On 10/15/2005, "John Novack" <[EMAIL PROTECTED]> wrote:
>Eric "ManxPower" Wieling wrote:
>
>> Andrew Kohlsmith wrote:
>>
>>> On Friday 14 October 2005 20:50, Eric "ManxPower" Wieling wrote:
>>>
Oddly enough, I believe it's mentioned in UPGRADE.txt.
>>>
>>> Care to tell us where? I just checke
On 10/15/2005, "J. Iddings" <[EMAIL PROTECTED]> wrote:
>I'm also having this issue. Everything seems to work, but it's an
>unnerving error. Any thoughts?
>
>Jimmy wrote:
>> I just upgraded my test Asterisk box to the latest CVS HEAD. "show
>> version" only shows "Asterisk CVS HEAD built by root..
I had a typo in my original mail - of course it's from junghanns.net,
not from digium, and so is the bristuff, that I downloaded.
I'm not searching for solutions on how to make it compile - I'm just
trying to find out if anybody succeeded in having it work on FC3 and
FC4, and if yes, if there are
--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:
> Are the devices at 200 and 310 set up to register with your asterisk?
Yes, they are registered and I can call them
__
Start your day with Yahoo! - Make it your home page!
http://www.yahoo.com/
Are the devices at 200 and 310 set up to register with your asterisk?
On Sat, 2005-10-15 at 11:42 -0700, Samy Antoun wrote:
Hi,
I have 3 SIP extensions, setup as follows:
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qua
Eric "ManxPower" Wieling wrote:
Andrew Kohlsmith wrote:
On Friday 14 October 2005 20:50, Eric "ManxPower" Wieling wrote:
Oddly enough, I believe it's mentioned in UPGRADE.txt.
Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt.
Sorry, it's in asterisk/configs/ext
I'm also having this issue. Everything seems to work, but it's an
unnerving error. Any thoughts?
Jimmy wrote:
> I just upgraded my test Asterisk box to the latest CVS HEAD. "show
> version" only shows "Asterisk CVS HEAD built by rootetc", with no
> date or version number. I downloaded this
Can someone tell me how I can test DID on analog lines ? I have the WCfxo clone and would like to try DID on it.
Can my telco (qwest) provide that service ?
-apu
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Asterisk-Users mailing list
Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.
In the true spirit of Open Source,
On Sat, 2005-10-15 at 14:29 +0200, Lars Dybdahl wrote:
> We have a QuadBRI ISDN card from Digium. We would like to make it work
> with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of
> bristuff from the Digium homepage fails, both the stable version with
> asterisk 1.0.9, and the e
Hi,
I have 3 SIP extensions, setup as follows:
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qualify=yes
This is the result of sip show peers:
Name/user Host Dyn Nat Status
200/200 192.168.1.150 D Unmonitored
This will actually be easy to fix. I'll post a patch along with
someother stuff shortly.
Darren
Darren Wiebe wrote:
That is true. It's just one of those things that is easier to leave
alone to avoid breakage in upgrades. It would be nice to get fixed
though
Darren Wiebe
[EMAIL PROTE
Finally(!), the answer is
in /etc/asterisk/zapata.conf add
'overlapdial=yes'
hope that helps someone with the same problem.
many thanks to [EMAIL PROTECTED] and gerold
On Mon, 3 Oct 2005 23:55:21 +0200
ChB <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> I have an asterisk-server (cvs-head from au
Actually, the QuadBRI card is not from Digium, but manufactured by Junghanns.net
Michael
2005/10/15, Lars Dybdahl <[EMAIL PROTECTED]>:
> We have a QuadBRI ISDN card from Digium. We would like to make it work
> with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of
> bristuff from t
Hi All,
I'm having problem with this phone.
Problems are regarding voicemail message alert on the phone.
---
handle_response: Host 'xxx.xxx.xxx.xxx' does not implement 'NOTIFY'
---
Can somebody help?
On the phone manual, is written that it can acept MWI, but... not mine!!!
Thanks!
--
.:FaberK:.
__
We use Cisco phones and we simply disabled call-waiting for those
lines. Don't know if that will help, but whatever soft/hardphone you
are using probably has a way to disable call-waiting.
Tom
On Oct 15, 2005, at 5:38 AM, João Paulo Antunes wrote:
Hi!
We have a big problem in our call cen
Search voip-info.org or google for features.conf.TomOn Oct 15, 2005, at 4:58 AM, Denis Vella wrote: Hi, We're trying to setup attended call transfer, but we have not been able to find the required configuration. Blind transfer works fine using the # key, but we don't like the fact that the tran
Tzafrir Cohen wrote:
On Sat, Oct 15, 2005 at 03:21:38PM +0100, Colin Martin wrote:
Hi,
I seem to be unable to get Asterisk to recognise the '#' key being
pressed to acknowledge an incoming call from a queue. No matter how many
times I press the key to acknowledge, the Asterisk server acts as
On Sat, Oct 15, 2005 at 03:21:38PM +0100, Colin Martin wrote:
> Hi,
>
> I seem to be unable to get Asterisk to recognise the '#' key being
> pressed to acknowledge an incoming call from a queue. No matter how many
> times I press the key to acknowledge, the Asterisk server acts as if I
> have n
kind of difficutl to help you if you dont provide the script or
relevant info about your php.ini configuration, stuff related to the
output buffer.
I used to do programming with phpagi class, but then i came up with something more simple and usefull for my purposes.
http://galileo.ivsol.net/scri
thats an Asterisk At Home mod.On 10/15/05, Tommy Denton <[EMAIL PROTECTED]> wrote:
Where can I get the maintenance panel for AMP? I have searched all over and cannot seem to find it.
Thank you in advance,
Tommy
___--Bandwidth and Colocation sponsored
Where can I get the maintenance panel for AMP? I have searched all over and cannot seem to find it.
Thank you in advance,
Tommy
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
htt
Hi,
I seem to be unable to get Asterisk to recognise the '#' key being
pressed to acknowledge an incoming call from a queue. No matter how many
times I press the key to acknowledge, the Asterisk server acts as if I
have not.
I have installed the ztdummy module, and it seems that Asterisk is
Kevin P. Fleming wrote:
Eric "ManxPower" Wieling wrote:
Sorry, it's in asterisk/configs/extensions.conf.sample
And the default is supposed to be 'on', so that it is backwards
compatible unless you turn it off (which is in the sample config file
so that new users will learn to build their
Eric "ManxPower" Wieling wrote:
Sorry, it's in asterisk/configs/extensions.conf.sample
And the default is supposed to be 'on', so that it is backwards
compatible unless you turn it off (which is in the sample config file so
that new users will learn to build their dialplans with it turned of
We have a QuadBRI ISDN card from Digium. We would like to make it work
with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of
bristuff from the Digium homepage fails, both the stable version with
asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD.
Has anybody here s
Hi!
We have a big problem in our call center: when an agent does an outgoing
call it can receive calls from the queues. The same happens if one agent
transfer a call for another agent... and the ringing tone while in a
call is puting the agents like crazy...
We have the hints working with li
Title: Message
Hi,
We're trying to
setup attended call transfer, but we have not been able to find the
required configuration.
Blind transfer works
fine using the # key, but we don't like the fact that the transferring
extension does not have any info on what happened to the
call.
Any
Hi List
I installed asterisk server and tried to transfer calls from asterisk to Clarent class 5 call manager.
The calls are passing through with out any problem but after 60 seconds the call get disconnected automatically.
Please help me to sort out this problem. Attaching here with my sip config
Andrew Kohlsmith wrote:
On Friday 14 October 2005 20:50, Eric "ManxPower" Wieling wrote:
Oddly enough, I believe it's mentioned in UPGRADE.txt.
Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt.
Sorry, it's in asterisk/configs/extensions.conf.sample
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