hello ALL,
My question is regarding SIP_CODEC.
1). How can I get which codec is used for this channel .
Ex: if incoming call to asterisk i want to know which codec is used for
this channel.
is there any way for printing codec in dial plan
2). How can I set codec for outbound
Hi all,
I have installed Trixbox 2.8.0.3 with TDM400p digium cards (2 fxs - 2
fxo). Before making security settings I need to backup all system. When
I click on Backup on System - Backup menu in admin panel anything come
up on screen. I tried to observe another way some users suggested
mondo for
On Wed, Dec 09, 2009 at 07:57:52AM +0100, Olivier wrote:
What's the output of:
lspci -v -nn -s 08:00.0
# lspci -v -nn -s 08:00.0
08:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network
Controller [HFC-8S] [1397:16b8] (rev 01)
Subsystem: Cologne Chip Designs
Hello users,
i am planning to forward my skype calls from skype to the asterisk registerd
skype.
The scenario is as follows.
i)SkypeUserA calls SkypeUserB
ii)SkypeUserB forwards his calls to SkypeUserC
iii)SkypeUserC sees he got call from SkypeUserA.
do i have a way to extract the
On Wed, Dec 09, 2009 at 08:54:13AM +0100, Olivier wrote:
Hi,
In /var/log/asterisk/full (Asterisk 1.6.2-rc6), I can see :
[Dec 8 15:02:17] VERBOSE[10283] config.c: == Parsing
'/var/spool/asterisk/voicemail/default/103/INBOX/msg.txt': [Dec 8
15:02:17] VERBOSE[10283] config.c: ==
srinivas Antarvedi wrote:
Hello users,
i am planning to forward my skype calls from skype to the asterisk
registerd
skype.
The scenario is as follows.
i)SkypeUserA calls SkypeUserB
ii)SkypeUserB forwards his calls to SkypeUserC
iii)SkypeUserC sees he got call from SkypeUserA.
I
2009/12/9 Tzafrir Cohen tzafrir.co...@xorcom.com
On Wed, Dec 09, 2009 at 07:57:52AM +0100, Olivier wrote:
What's the output of:
lspci -v -nn -s 08:00.0
# lspci -v -nn -s 08:00.0
08:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network
Controller [HFC-8S]
Daniel Stefanus wrote:
I want to rebuild my mixmonitor file.But this time I just want the
recording is from the time when the client answer the call,not from the
beginning. Anybody can help?
Daniel
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Hello,
I am now working on a project which need data modem calls passing trough
asterisk with a TDM card. But i found out (as Mr Greg woods) that it was
impossible. So know i am trying doing this using an E1/T1 card (TE110P). I
am still working on it so we can help each other to find a solution to
2009/12/9 Tzafrir Cohen tzafrir.co...@xorcom.com
On Wed, Dec 09, 2009 at 08:54:13AM +0100, Olivier wrote:
Hi,
In /var/log/asterisk/full (Asterisk 1.6.2-rc6), I can see :
[Dec 8 15:02:17] VERBOSE[10283] config.c: == Parsing
I have Asterisj 1.4.26.`1, and Dahdi 2.2.0.2.
I restarted for no good reason (I was playing around), but it did worry me
that if Dahdi crashed while Asterisk was running that not only Dahdi and
Asterisk would crash, but the whole machine too.
Mike
-Original Message-
From:
On 12/08/09 20:43, Christian Victor wrote:
2009/12/8 Joseph syscon...@gmail.com:
After pressing *1 console is not showing anything indicating that the call
is being recorded:
-- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0,
transfer) in new stack
? ? --
Just a guess, but the connection probably went from full to half duplex.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: Tuesday, December 08, 2009 8:54 PM
To: Asterisk Users Mailing
On Wed, 2009-12-09 at 05:34 -0600, mos...@infolog.mr wrote:
I am now working on a project which need data modem calls passing trough
asterisk with a TDM card. But i found out (as Mr Greg woods) that it was
impossible.
Just to be clear about this: don't use me as an authoritative reference
snip
Just a guess, but the connection probably went from full to half duplex.
/snip
Full vs. Half duplex networking would NOT cause half duplex phone calls.
-Dave
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Hello everybody,
I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used
as PSTN gateway to asterisk in a small office. Everything works just
fine, except that sometimes, and it seems that only for long incoming
calls, the IVR menu appears on the middle of the call(like a three
This probably isn't a good guess either, but could OP have just 5060 open
and not the other ranges * needs to run the call? Since he has one-way
connection, I'm guessing not...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
HaHa!. That is so funny, made me splurt my coffee over the keyboard.
lol
Julian
2009/12/9 Danny Nicholas da...@debsinc.com:
Just a guess, but the connection probably went from full to half duplex.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
I found a solution myself I want to share it.
The pjsua SIP client from http://www.pjsip.org does the trick.
You can run it parallel to the asterisk server, ring for a while and hang up
then.
You just need to change one line in the source code to run it in background:
In
Hi list:
I'm having problems, with a AEX800P card when plugin on HP ML115 G5
Server, when i load the mdule (wctdm24xxp), it loads with error con
dmesg, that say that is a kernel bug of invalid OPCODE 000 or
something like that.
If i plug the card on SUN Server, i don't have problems with
Thanks a lot for that info Christian.
Date: Tue, 8 Dec 2009 20:14:02 +0100
From: Christian Victor christ...@victormedia.de
Subject: Re: [asterisk-users] Sangoma A101DE with Dell PE 2850
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Tzafrir,
Would you say latest trunk (revision 7672) includes a fix for those OctoBRI
boards ?
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To UNSUBSCRIBE or update options visit:
On 12/09/2009 05:39 AM, Mike wrote:
I have Asterisj 1.4.26.`1, and Dahdi 2.2.0.2.
I restarted for no good reason (I was playing around), but it did worry me
that if Dahdi crashed while Asterisk was running that not only Dahdi and
Asterisk would crash, but the whole machine too.
Mike,
Could
I am revising our DialPlan strategy for our Asterisk system (1.4.2) and
looking for some info on 'best practices' for this. Here's what I'm
trying to do:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
-
Looking for IP Phone recommendation to be used with FreePBX. Approx. 50 IP
Phones, the FreePBX behind router. So good NAT support needed. FreePBX is
installed from the iso image available in AsteriskNOW website.
Sorry for the cross post, hoping that the 2 lists provide 2 different
perspective.
If you add option 6 to the menu for the first position and use the
read command for the 2 last position and use a second line that looks
something like:
exten 6,n,Dial(SIP,6${ENTERED_NUMBER},20,t)
it should work.
The {ENTERED_NUMBER} should be the variable filled with the read
command.
Have you tried...
exten = 6XX,1,Dial(SIP,${EXTEN},20,t)
--
Jarrod Lash, jar...@fed-com.com
Federated Communications, LLC.
www.fed-com.com
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368
On Wed, Dec 9, 2009 at 12:53 PM, Myles Wakeham my...@techsol.org wrote:
I am
On Wed, Dec 09, 2009 at 05:59:36PM +0100, Olivier wrote:
Tzafrir,
Would you say latest trunk (revision 7672) includes a fix for those OctoBRI
boards ?
The only change was in dahdi-tools . As could be clearly seen from the
output of lspci, wcb4xxp already handles this device.
--
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that extension.
Yes, I can hear the tones as I'm pressing the *1, it has to do with the
timing, pressing the buttons with two finger is quicker than with one.
Can someone provide where is the timing Francesco in previous post
mentioned.
It's featuredigittimeout (in milli-seconds). Try
featuredigittimeout =
The SPA-3000 is notorious for falsely detecting DTMF tones in regular
voice, and when it thinks it hears DTMF, it will produce a short
real DTMF tone that's only audible to the SIP side of the device, not
the PSTN side, or out of band SIP DTMF message (dependent on how you
have the device setup).
I'm running * 1.4 and can successfully restart asterisk from the command
line with:
/usr/sbin/asterisk -r -x restart gracefully
However, I have a cron job that tries to restart asterisk and gets this
error:
No such command 'restart gracefully' (type 'help restart gracefully' for
other possible
On 12/09/09 13:40, Kai-Uwe Jensen wrote:
Yes, I can hear the tones as I'm pressing the *1, it has to do with the
timing, pressing the buttons with two finger is quicker than with one.
Can someone provide where is the timing Francesco in previous post
mentioned.
It's featuredigittimeout (in
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the command
line with:
/usr/sbin/asterisk -r -x restart gracefully
However, I have a cron job that tries to restart asterisk and gets this
error:
No such
Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.
Not sure of what release you're testing on.
See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad...
Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.
Not sure of what release you're testing on.
See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad...
Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.
Not sure of what release you're testing on.
See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad...
Andrew Hakman wrote:
The SPA-3000 is notorious for falsely detecting DTMF tones in regular
voice, and when it thinks it hears DTMF...
Thanks Andrew. This is true, we do have false DTMF detections/playback
(Lowering RX gain really helps on this). However this does not seem to
be related. The
Don't forget that many routers treat the designated private address space
differently because it assumes the device is being implemented as a border
router. In this configuration they block most traffic unless you
specifically set rules to permit traffic to flow.
-dbc.
Hey folks, I'm from Brazil and I have the following doubt. May I use
an asterisk box with some cards to act as an PSTN simutator between a
little amount of sites? I will start to think about it in a voice lab
for my studies.
Kind regards,
Vitor
--
Vitor Afonso Strabello
MSN:
Depends on how authentic you want your simulation to be.
On 12/09/2009 07:49 PM, Vitor Afonso Strabello wrote:
Hey folks, I'm from Brazil and I have the following doubt. May I use
an asterisk box with some cards to act as an PSTN simutator between a
little amount of sites? I will start to
On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys voi...@gmail.com wrote:
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call was originated, for
example client's_number - Sales. This problem appears when one member
can
On Thu, Dec 10, 2009 at 2:54 AM, Atis Lezdins a...@iq-labs.net wrote:
On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys voi...@gmail.com wrote:
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call was originated, for
Warren Selby wrote:
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca
mailto:supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the
command
line with:
/usr/sbin/asterisk -r -x restart gracefully
I have the following cron job:
Doug Lytle wrote:
Warren Selby wrote:
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca
mailto:supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the
command
line with:
/usr/sbin/asterisk -r -x restart gracefully
I
But my command works from a bash command line...so something is fishy!
I want to solve this mystery
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Wednesday, December 09, 2009 8:27 PM
To:
Interesting...I'll try that. Thanks
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese
Sent: Wednesday, December 09, 2009 8:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Can't restart asterisk from
Keep in mond that cron usually has a very abbreviated environment for
security reasons - you may need to set the PATH or other environment
variables in the crontab to get it to work.
Billk
On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote:
Interesting...I'll try that. Thanks
But the error message in my log shows the error to be from asterisk, so I'm
guessing I'm sending a parameter incorrectly to asterisk - which fits with
the no quote theory
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hi!
I saw your profile and would like to get to know you better.
Im looking for open, adventurous people, in my area, but we can start here.
Email me back at maris...@email-chatting.com .
Muah!
Marishka ;-)
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Hello,
We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would
like to use the sip,extensions and voicemail in realtime mode.
Where can we find the database tables structure for these versions?
Thanks,
Andreas
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You should replace the single quote with double quote.
--Original Message--
From: Michelle Dupuis
Sender: asterisk-users-boun...@lists.digium.com
To: 'Asterisk Users List'
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't restart asterisk
On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote:
Hello,
We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would
like to use the sip,extensions and voicemail in realtime mode.
Where can we find the database tables structure for these versions?
Thanks,
Andreas
This is
Thanks Fred,
I'm actually there, but I was wondering if the tables there are up to
date and if any changes took place. I see all kinds of comments about
changes.
Original Message
Subject: Re: [asterisk-users] Realtime Database Tables
From: Fred Posner f...@teamforrest.com
I had double quotes originally - and that didn't work
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E.
Rodríguez
Sent: Wednesday, December 09, 2009 10:14 PM
To: Asterisk Users List
Subject: Re:
Yes, but if asterisk cant find some of its components due to abbreviated
path or ...
Just run a cron that prints the results from env and compare and see
if there is something obvious - there may also be privilege issues
BillK
On Wed, 2009-12-09 at 22:32 -0500, Michelle Dupuis wrote:
I had
On Wed, 9 Dec 2009, Michelle Dupuis wrote:
I'm running * 1.4 and can successfully restart asterisk from the command
line with: /usr/sbin/asterisk -r -x restart gracefully
However, I have a cron job that tries to restart asterisk and gets this
error:
No such command 'restart gracefully'
On Wed, Dec 09, 2009 at 03:28:19PM -0600, Warren Selby wrote:
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the command
line with:
/usr/sbin/asterisk -r -x restart gracefully
However, I have a cron
hello ALL,
i have one virtual machine creating via XEN we running conference on that
using SIP channel
everything is fine except recording a .wav file is 2x times faster that
original voice
is there any setting regarding that to improve it?
regards
Dhaval
2009/12/9 Michelle Dupuis supp...@ocg.ca
I'm running * 1.4 and can successfully restart asterisk from the command
line with:
/usr/sbin/asterisk -r -x restart gracefully
However, I have a cron job that tries to restart asterisk and gets this
error:
No such command 'restart gracefully' (type
On Thu, Dec 10, 2009 at 08:02:22AM +0100, Olivier wrote:
2009/12/9 Michelle Dupuis supp...@ocg.ca
I'm running * 1.4 and can successfully restart asterisk from the command
line with:
/usr/sbin/asterisk -r -x restart gracefully
However, I have a cron job that tries to restart asterisk
On 12/10/2009 02:07 AM, Tzafrir Cohen wrote:
Why would one want a daily sabotage of the system in the first place?
An apt question.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
2009/12/10 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Dec 10, 2009 at 08:02:22AM +0100, Olivier wrote:
2009/12/9 Michelle Dupuis supp...@ocg.ca
I'm running * 1.4 and can successfully restart asterisk from the
command
line with:
/usr/sbin/asterisk -r -x restart gracefully
I am new to the list and wanted to get the professionals here input on
Switchvox 305 Appliance ?
List price is 4k, ouch! Is there a better cost-effective way ?
Also feedback (neg/pos) about this appliance.
-mike
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