Hi Greg
Thanks for the input. Just to make sure I dont break my system, could I run
the proposed changed by you?
exten = s,n(a42),Dial(${${HuntMember}}${ds}) produces this - -- Executing
[EMAIL PROTECTED]:38] Dial(Zap/29-1, IAX2/1101|20|trM(auto-blkvm)) in new
stack
so the mod would
You missed a slash.
v
exten = s,n(a42),Dial(${${HuntMember}*/*${EXTEN}}${ds})
^
Louwrens Benadé wrote:
Hi Greg
Thanks for the input. Just to make sure I don’t break my system, could
I run the proposed changed
Problem:
When I have more than one IAX2 connection (on server zuiderven), I have
problems in receiving calls from IAX peers except for the first in the
list as seen by the iax2 show peers command.
In my tests it showed that by removing one by one the entries from the
iax.conf file in the order as
Resolved!
The problem was with sip.conf file. I had to comment the lines
allow=alaw
allow=ulaw
This made the trick..
I am trying to get the mysql database connection from my asterisk box.
Installed the asterisk-addons-1.4.4 version on the same box where asterisk and
mysql is installed.
Hi all,
Do some one experiencing running jabber applications (jabberstatus...) in
asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I
got such result.
IBM*CLI help jabber
No such command 'jabber'.
IBM*CLI help jabberstatus
No such command 'jabberstatus'.
Any one can
hello everyone,
I have a trixbox server with an E1 card(not digium).It connects to an
AVAYA pbx use E1. It works fine.But when i change the E1 card to digium
TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension
B(on avaya),A must wait longtime before B start to
Mark Quitoriano wrote:
On Sat, Feb 16, 2008 at 12:31 AM, Faruk Kasumovic
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Johansson Olle E wrote:
Hi Mark!
13 feb 2008 kl. 23.42 skrev Mark Quitoriano:
Is it possilble for a single context to have
Lol, thanks Rob :-)
I just saw that, its working properly now. Although, in the Trixbox
environment I had to use ${DID}, as ${EXTEN} just produced another s.
Now I just have to hope that my ugly FaxDispatch hack works as well
Just for those interested, heres what it looks like:
On Tue, 2008-02-19 at 17:59 +0800, [EMAIL PROTECTED] wrote:
hello everyone,
I have a trixbox server with an E1 card(not digium).It connects
to an AVAYA pbx use E1. It works fine.But when i change the E1 card to
digium TE220B,there is a problem. When sip extension A(on trixbox)
call PSTN
Hi,
I have register a sip user to sip server. I can see after registration * is
sending periodic SIP Request: OPTIONS messages to server. but it's not
getting back any response that should be SIP 200/OK as the documents say.
3130.299707 192.168.2.113 - 58.ab.cd.ef SIP Request: OPTIONS sip:
I read from the forums, that if i build mysql the problem will be resolved. As
i get the similar warning message for MeetMe().
How to build mysql or MeetMe manually??
Regards,
Naveen.Palani
- Original Message -
From: Naveen Palanimailto:[EMAIL PROTECTED]
To:
Hi
I am having a problem with proper registration to asterisk through IAX. The
peer (which is Iaxmodem) suppose to register to the server each 60 sec and
it is doing so, but the server is aware of the registration only for first
ten seconds and after that time ipaddr and regsec fields in database
Hi Naveen
If you compiled * from source, what you would need to do is to satify the
dependencies for these modules. In the src directory, go ‘make menuselect’ and
find the app, if you see ‘XXX’ next to it, it means it cannot compile due to
unresolved dependencies. At the bottom of the
On Tue, 2008-02-19 at 17:00 +0500, ast guy wrote:
Is it useful, harmful, what If I want to disable this periodic
request?
It can be useful, as Asterisk will keep track of whether or not that
peer is responding. You can turn off these SIP OPTIONS packets from
being sent by setting qualify=no in
I used mib2c on the ASTERISK-MIB in this following manner:
mib2c asterisk
used UCD-SNMP type to obtain the asterisk.c and asterisk.h file.
But there is no code from which I can learn the way the ASTERISK-MIB works.
Can someone please tell me from where can I get the appropriate .c and .h
file
On 2/19/08, Alex Balashov [EMAIL PROTECTED] wrote:
Or, you can write your own scripts to generate calls via the Manager
API, or use Asterisk call files (see voip-info.org on this topic).
But, all other things being equal, it is probably preferred to use some
sort of testing framework of the
I've upgraded my server from asterisk 1.2 to 1.4.18 and CDR has started to
log two lines for each failed call (NO ANSWER) instead of one per call as it
was done in 1.2.
In my extensions.conf, call flow starts in default context and then jumps to
a macro where is then dialed the destination. When
Is there any way that I can have an admin user hit * and then Mute all other
users in a meetme conference? Sort of a moderator function?
I know it can be done with MeetMeAdmin, but as I see it that requires a
separate extension to dial, unless I've got the logic wrong?
If it can be done in a
In article [EMAIL PROTECTED],
Jeremy Mann [EMAIL PROTECTED] wrote:
Is there any way that I can have an admin user hit * and then Mute all other
users in a
meetme conference? Sort of a moderator function?
I know it can be done with MeetMeAdmin, but as I see it that requires a
separate
Alex Balashov [EMAIL PROTECTED] schrieb im Newsbeitrag
news:[EMAIL PROTECTED]
Seysan wrote:
Hello,
What is the Differences between Asterisk 1.4 and Asterisk 1.6 ?
Also I mean what has made it to be in a separate Branch ?
There are release notes that speak to this.
Where? And is
To you extensions.conf gurus, I'd like some help on having a button/feature to
turn on/off system wide call forwarding.
I need the phone system to forward calls received, after the feature is
activated, to an answering service.
Calls received are on a PRI. I need all DIDs forwarded once the
Hi, i'm new to the list, this is my first post here.
I have a UMTS module that can be used like a modem and associated with a SIM
card (on a simple board) can make phone and data call towards a real phone.
Basically is a sort of rough phone that i configure and use by AT commands.
It can handle
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an interface),
any recommendations?
Anton Krall
Direccion General
Intruder Consulting
A Division of
At 06:49 2/19/2008, Peter Nabbefeld wrote:
Alex Balashov [EMAIL PROTECTED] schrieb im Newsbeitrag
news:[EMAIL PROTECTED]
Seysan wrote:
Hello,
What is the Differences between Asterisk 1.4 and Asterisk 1.6 ?
Also I mean what has made it to be in a separate Branch ?
There
Perfect! Thanks.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield
Sent: Tuesday, February 19, 2008 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MeetMe Admin Functions
In article [EMAIL PROTECTED],
Jeremy Mann
Hi All;
What is the command that I can type it to determine
the zaptel version, also where I can find information
about the compatibility matrix that I have to follow
it for my asterisk, zaptel and kernel version to be
sure that I am installing compatible parts?
Any help?
Regards
Bilal
Cavalera Claudio Luigi [EMAIL PROTECTED] writes:
Hello,
I'm experimenting with Asterisk and MySQL.
Up to know I've just put iax.conf in a MySQL database and it seems to
work: when a Iax2 client registers the corrispondent row in db is
updated. Good.
However when I have many asterisk boxes
On Tue, Feb 19, 2008 at 10:16:26AM -0800, bilal ghayyad wrote:
Hi All;
What is the command that I can type it to determine
the zaptel version,
On a non-ancient kernel:
cat /sys/module/zaptel/version
Failing that, you can check the version of the module that will probably
get loaded:
On 18/02/2008, Razza [EMAIL PROTECTED] wrote:
Thanks every one, having looked at
http://www.cisco.com/warp/public/788/products/1750-vic-issues.html and
http://www.cisco.com/en/US/products/hw/routers/ps259/products_tech_note09186a00800e73f6.shtml
It appears to support a single BRI I need
Okay, so I've been toying around on the Rolm side, and still getting nothing.
Took another look on Asterisk, finally figured out where the debugging could be
enabled on the console, and finding a lot of interesting things.
Running 'dmesg' simply shows the entire buffer is flooded with 'PCI
On 13:49, Tue 19 Feb 08, Peter Nabbefeld wrote:
Alex Balashov [EMAIL PROTECTED] schrieb im Newsbeitrag
news:[EMAIL PROTECTED]
Seysan wrote:
Hello,
What is the Differences between Asterisk 1.4 and Asterisk 1.6 ?
Also I mean what has made it to be in a separate Branch ?
On Tue, Feb 19, 2008 at 01:49:58PM +0100, Peter Nabbefeld wrote:
Alex Balashov [EMAIL PROTECTED] schrieb im Newsbeitrag
news:[EMAIL PROTECTED]
Seysan wrote:
Hello,
What is the Differences between Asterisk 1.4 and Asterisk 1.6 ?
Also I mean what has made it to be in a separate
Like 15 lines of php and html?
?php
$fn = /etc/asterisk/extensions.conf;
if ($_REQUEST['action'] == write $_REQUEST['contents'] != )
{
rename($fn, $fn...date(U));
$fp = fopen($fn, wt);
fwrite($fp, $_REQUEST['contents']);
fclose($fp);
}
?
form
Just out of curiosity, why PHP?
Atis Lezdins wrote:
On 2/19/08, Alex Balashov [EMAIL PROTECTED] wrote:
Or, you can write your own scripts to generate calls via the Manager
API, or use Asterisk call files (see voip-info.org on this topic).
But, all other things being equal, it is probably
On Sun, Feb 17, 2008 at 01:50:32PM -0600, John Hawley wrote:
Hi,
Would anyone have a clue on this issue?
I'm running asterisk 1.4.13. Trying to get a WAV voicemail file
attachment sent to my email address.
Voicemail is working fine. Email notification of a new message works
fine.
Jared Smith wrote:
I've always done this by setting the MONITOR_EXEC channel variable to
point to an external program that takes care of moving the recording to
the proper location so that it can be accessed by the user who made the
recording. I'll bet if you search for MONITOR_EXEC and
Hi All;
I am facing sometimes (not always) a problem in
detecting the entered digits when calling from outside
to the fxo.
I noticed that a dupilcation happens: for example if I
entered 1 then it detect it 11, any help?
I tried relaxdtmf = yes but still no result.
Any help?
Regards
Bilal
On Tue, 2008-02-19 at 20:59 +, Ben Willcox wrote:
That gets me halfway there, but what I'm wondering about is the process
of moving of the recording to the correct place - i.e. should my
external program do the following:
1) Check the users voicemail directory for existing message
Okay, some more interesting tidbits to throw out incase someone has run into
this before.
I've found out that th D100P has been EOL'ed by Digium due to it being a bit
weird with certain systems, and I suspect my HP Proliant DL385 server may be
one of those. Anyone used this card on such a
Michelle Dupuis escribió:
I have setup hylafax today, along with iaxmodem. I'm just starting
the debugging process and see the following message every 60 seconds:
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry:
Restricting registration for peer 'iaxmodem0' to 60 seconds
There's a #define macro in channels/chan_iax.c that you can modify to make this
forced value higher. Just open it up in your favourite editor and search for
'60' and you'll find it.
Now if there's an easier way than having to change a source-level macro, I'm
all ears...
Cheers!,
--jkinard
I have setup hylafax today, along with iaxmodem. I'm just starting the
debugging process and see the following message every 60 seconds:
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry:
Restricting registration for peer 'iaxmodem0' to 60 seconds (requested 300)
Can someone
The system won't boot at all if the tor2.ko file exists. Period. It
doesn't matter what is in the startup scripts. Even if I DELETE the
startup scripts, it still crashes until I go in with a rescue CD and
delete tor2.ko from the modules directory for the kernel.
It's a full kernel panic...
Hi folks,
I'm running Asterisk 1.4.18 on Mac OS X. I'm using Broadvoice to
connect external numbers with this system. So, what I'm dealing with
here are inbound calls.
The problem is that I'm not able to get DTMF in. I am behind NAT, but
I've made sure that the IT folks here have opened
Hi all, I am a huge fan of Sangoma cards after having many problems with
digium cards and then switching to sangoma cards and them giving me
excellent support with excellent results.
That being said, they are also alot more money than the Rhino cards and
my friend currently has 1 digium 4 fxo
I have just been given the answer -
exten = *44,1,Answer
exten = *44,n,Noop(Check Callforward)
exten = *44,n,Set(_MISDN_KEYPAD=*#24#)
exten = *44,n,Dial(mISDN/1/)
exten = *44,n,Hangup
Where the line 'Set(_MISDN_KEYPAD=*#24#)' is where you set your code.
later,
PaulH
On Tue, 2008-02-19 at
On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote:
Hi all, I am a huge fan of Sangoma cards after having many problems with
digium cards and then switching to sangoma cards and them giving me
excellent support with excellent results.
I would recommend that you give the Digium
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de febrero de 2008 01:35 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk
I increased that to 300, and restart *. The error still pops up...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rodrigo Gonzalez
Sent: Tuesday, February 19, 2008 6:13 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Restricting
Wow...where did you get that answer?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Paul Hales
Sent: Tuesday, February 19, 2008 8:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] ISDN2 facility code...
I have just been given the
From the local Digium reseller.
Not something you would have guessed though, is it?
I should probably enter this onto voip-info.org
later,
PaulH
On Tue, 2008-02-19 at 21:01 -0500, Michelle Dupuis wrote:
Wow...where did you get that answer?
-Original Message-
From: [EMAIL
On Feb 19, 2008 10:44 AM, Anton Krall [EMAIL PROTECTED] wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an interface),
any recommendations?
You mean, something like this :
Hello,
I updated my AJAX GS/* phonebook. It now actually has Asterisk integration with
incoming call-notification, a searchable call history and clickable BLFs.
The BLFs can monitor specific extensions (SIP/phone - may work with other
channels than SIP if the AMI generates the appropriate
Kevin P. Fleming wrote:
sean darcy wrote:
That is, is port 1 = channel 1 and slot 1?
Yes, they are. However, 'UNCONFIGURED' means you haven't run ztcfg yet,
so Asterisk cannot use the channels.
Thanks.
sean
___
-- Bandwidth and Colocation
using asterisk(A) over iax to another asterisk server(B) which connects
to pstn over pri.
Doesn't B have translate to ulaw whatever goes out to the pstn, so
therefore shouldn't A choose ulaw as the iax codec to B? That way
there's no loss translating from {gsm, ildc, etc} to ulaw on the B
Tilghman,
Could you clarify what you mean when you say you added usleep(1) to the end
of the manager thread? I do not have enough experience to follow what you're
saying.
Are you talking about adding this command to the end of the manager.conffile?
- Mark
Hello All
We need to connect our client's offices located in Dubai and
Pakistan. Suggest us some economical solution.
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN: [EMAIL
On Tuesday 19 February 2008 22:20:06 Mark Greene wrote:
Could you clarify what you mean when you say you added usleep(1) to the
end of the manager thread? I do not have enough experience to follow what
you're saying.
I added the comment about what I did to fix it in the source, for those who
Hi,
I need to subscribe and use several Polycom-Kirk 5020 handsets along
non-Polycom-Kirk handsets on a one-cell Polycom-Kirk 600/3 base station.
Has anyone tried this ?
Which values did you pick for Subscription mode (with or without Account
Code) and IPEI ?
Regards
OK thanks for the effort.
What's a way to look for IRQ misses in linux?
On Feb 20, 2008 12:33 AM, Tilghman Lesher [EMAIL PROTECTED]
wrote:
On Tuesday 19 February 2008 22:20:06 Mark Greene wrote:
Could you clarify what you mean when you say you added usleep(1) to the
end of the manager
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