I'm no longer on the DEV mailing list, but:
# svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk
svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist
http://svn.digium.com/svn/asterisk/branches/
--
/Nick
___
Hi all,
I have a TDM400 with all FXO on it. When I make an outgoing call, I
can hear callee but callee claims the volume is too low so that he/she
can't hear very clear. Can I adjust to increase the volume in callee
side? Is it increase the value of txgain can solve the problem?
ango
Ciao Olle, Tilghman,
How could I do it using the hint mechanism?
Just create a module that subscribes to every single device and when
the state
changes, your callback will get an event with the device name that
changed.
You could then update your database with an SQL query (or
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nicholas Blasgen wrote:
I'm no longer on the DEV mailing list, but:
# svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk
svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist
On Jan 16, 2008 9:18 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
can't hear very clear. Can I adjust to increase the volume in callee
side? Is it increase the value of txgain can solve the problem?
Ango,
If you search around for zaptel txgain you'll probably find a lot of
info about tweaking
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rilawich Ango wrote:
Hi all,
I have a TDM400 with all FXO on it. When I make an outgoing call, I
can hear callee but callee claims the volume is too low so that he/she
can't hear very clear. Can I adjust to increase the volume in callee
side?
The TE121 is a PCI Express card (TE122 is standard PCI, 2.2 if I'm not
mistaken).
Gal Barak
Tech support
Atelis PLC
2008/1/16, Guilherme Loch Waltrick Góes [EMAIL PROTECTED]:
What's the difference between the TE121 and TE122. I read the description
on Digium's site and it isn't clear to me.
Hi,
I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to
make zaptel working...
OS is gentoo linux 2006.1
Hardware:
-
:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device 8085:0003
Flags:
On Wed, Jan 16, 2008 at 04:18:39PM +0800, Rilawich Ango wrote:
Hi all,
I have a TDM400 with all FXO on it. When I make an outgoing call, I
can hear callee but callee claims the volume is too low so that he/she
can't hear very clear. Can I adjust to increase the volume in callee
side? Is
On Wed, Jan 16, 2008 at 03:55:08PM +0500, ast guy wrote:
Hi,
I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to
make zaptel working...
OS is gentoo linux 2006.1
Hardware:
-
:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN
Hi!
I'm building an application which allows to dial via the Asterisk
Manager Interface using the originate command. There should be an
optional conferencing feature.
The manager commands are basically:
-
action: login
username: sdjklgdsjg
secret: xxx
events:
What's the difference between the TE121 and TE122. I read the description on
Digium's site and it isn't clear to me.
Best regards,
--
Guilherme Loch Góes
Visite nossa loja virtual: http://www.shopvoip.com.br
Notícias e Fórum sobre VoIP com software livre:
http://www.asteriskexperts.com.br
Hello!
Guilherme Loch Waltrick Góes wrote:
What's the difference between the TE121 and TE122. I read the description on
Digium's site and it isn't clear to me.
Best regards,
The only one difference is interface: one of them have PCI and other
have PCI-Express.
--
Best regards,
Igor A.
Hi all
This is rahul i am using asterisk 1.4.17 with degium TE120p card on PRIE.
I have configured everything card but there is a problem coming
asterisk is dialing _98 is not dialing _99 showing
Everyone is busy/congested at this time
zap/1-1 CHANISUNAVAIL.
I have tried all
Please expliain more, show us your extensions.conf.
On Jan 16, 2008 9:51 AM, Rahul Yadav [EMAIL PROTECTED] wrote:
Hi all
This is rahul i am using asterisk 1.4.17 with degium TE120p card on PRIE.
I have configured everything card but there is a problem coming
asterisk is dialing _98
G'day.
Has anyone here used the Dualphone SIP products (their LAN range)
together with Asterisk sufficiently to comment on them?
http://dualphone.net/
https://www.dualphone.com.au/product_info.php/products_id/51
I am interested, specifically, in these questions:
* are they generally
On 1/16/08, Franz Schwartau [EMAIL PROTECTED] wrote:
Hi!
I'm building an application which allows to dial via the Asterisk
Manager Interface using the originate command. There should be an
optional conferencing feature.
The manager commands are basically:
-
Good Day All,
Is it possible to put backup route in asterisk dial plan? fro the example if
the first carrier disconnect the call with Congestion or Circuit busy then
asterisk can dial another carrier?
I did the following but it is not working as i need to dial the second one only
on
On Jan 15, 2008 8:12 PM, John Millican
[EMAIL PROTECTED] wrote:
Hello all,
Was hoping to get a sanity check along with a question. Below is the
output from top run with normal defaults, except to show both CPU's, on
a SuSE 10.2 box with Asterisk v1.4.15.
[snip massive hardware spec]
We have
Hi All;
I did an IP Trunk using IAX between two Asterisk
boxes, now Asterisk A can send a call for B but B
refuse it. The IAX type was configured to be friend
in the iax.con for Asterisk A and B, is there any
thing else need to be done to let B accept the call
from A?
Also, I used an static IP
Hi All;
Did anyone tried to use dns name or ddns name with
host (host=abc.www.com) and it worked fine?
Regards
Bilal
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
Dear All,
Thank you for taking the time to read my message. I have just installed
Asterisk Now, and it seems to be up and running with no issues on my system.
The problem I am facing, is that I cannot find anywhere in the web
interface, to assign phones. I have a CISCO IP phone 7910 series,
Hi AK;
I would like to ask a question: where is the problem
if u record the prompted messages in ur voice and as u
need? Does not work?
Also, if that the situation: how can I determine the
needed voicemail language? For example I need ARABIC
language, so what should I do to have arabic prompts?
Well can you offer some explanation why T.38 faxing worked for months
and then one day stopped working?
Generally it is because some one or some process did one or more of the
following:
1. Updated firmware on the ATA
2. Updated software on the server
3. Changed a configuration setting
4. Let
I have install a Asterisk 1.4.9 with Centos, a TDM400P (4 Analog Lines) my
problem is one o two day a week one of the lines have a lot of noise, I i
cant place a call outside.I need to reboot the server to get the lines
again. Do you know is it's another way to check the lines o reset the
Hello,
A user who uses my Asterisk made me part of a worry about listening to his
voicemails. He has received 4 voicemails on January 3, respectively at 3H00
pm, 3H36 pm, 3H41 pm and 4H40 pm. He has received notifications by e-mail at
these times.
On first listen to his messages, at 8.00 pm,
Change the priority of the second dial() to 4.
Regards,
On Jan 16, 2008 11:42 AM, Abdul [EMAIL PROTECTED] wrote:
Good Day All,
Is it possible to put backup route in asterisk dial plan? fro the example
if the first carrier disconnect the call with Congestion or Circuit busy
then asterisk can
On Tue, Jan 15, 2008 at 01:14:42PM +, Christian Pinedo wrote:
hi,
I'm trying to configure a Cisco IP Phone 7911G in order to work with
Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a
TFTP server. All seems ok but a file that is downloaded :
I would suppose that the time on the asterisk system is not the time
that he is using. Other than that, you should really be collecting logs.
David Florella wrote:
Hello,
A user who uses my Asterisk made me part of a worry about listening to
his voicemails. He has received 4 voicemails on
Thank you very much, that was a new angle I hadn't thought of time to
investigate a little more :). The joys of learning new things :)
- Christian
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC
Sent: 16.
Now that you have your 7911g phone up running, would you mind checking
the audio quality when leaving a voicemail for on another local asterisk
user from this phone? I have a 7911g and I hear loud audio taps from the
phone. The 7961g phone doesn't have this issue. I'm just trying to
rule out the
Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3
Upgraded this morning, now PRI channels are unstable as hell. After about 5
minutes all asterisk commands on the console refuse to respond, attached is the
debug log right before and after the lock-up, IT occurred between 9:18 and
9:20 AM at
Yes, I use it and got no problems.
On Jan 16, 2008 11:47 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
Did anyone tried to use dns name or ddns name with
host (host=abc.www.com) and it worked fine?
Regards
Bilal
Igor A. Goncharovsky wrote:
Hello!
Guilherme Loch Waltrick Góes wrote:
What's the difference between the TE121 and TE122. I read the description on
Digium's site and it isn't clear to me.
Best regards,
The only one difference is interface: one of them have PCI and other
have
Hello
Before I bother calling a PHP script through AGI just to read a number
and rewrite the CID name... I was wondering if Asterisk could be
configured so that DB() uses a SQL server instead of the usual
BerkeleyDB?
;rewrite CIDNAME if found in DB
exten =
On Jan 16, 2008 10:25 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3
Upgraded this morning, now PRI channels are unstable as hell. After about
5 minutes all asterisk commands on the console refuse to respond, attached
is the debug log right
On Jan 16, 2008 8:46 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
I did an IP Trunk using IAX between two Asterisk
boxes, now Asterisk A can send a call for B but B
refuse it. The IAX type was configured to be friend
in the iax.con for Asterisk A and B, is there any
thing else need
Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are behind NAT... I'd like
to try IAX to
Jeremy Mann wrote:
Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3
Upgraded this morning, now PRI channels are unstable as hell. After about 5
minutes all asterisk commands on the console refuse to respond, attached is
the debug log right before and after the lock-up, IT occurred
On Wed, 16 Jan 2008, Vincent wrote:
Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are
On Wed, 16 Jan 2008, Vincent wrote:
Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are
On Wednesday 16 January 2008 10:02:12 Vincent wrote:
Before I bother calling a PHP script through AGI just to read a number
and rewrite the CID name... I was wondering if Asterisk could be
configured so that DB() uses a SQL server instead of the usual
BerkeleyDB?
No, it cannot. You could use
Dave Fullerton wrote:
If you want to know what a card's capabilities are you're better off
just memorizing each part number. Maybe there's a scheme I'm just not
capable of understanding here.
We gave up (intentionally) on trying to have model numbers that
reflected all the capabilities of
Did you look at the trace I send you in email? Because in each request
there are two IN IP lines I think Asterisk should only interpret the
first one,
On Jan 16, 2008 2:40 AM, Johansson Olle E [EMAIL PROTECTED] wrote:
16 jan 2008 kl. 04.43 skrev Andrew Joakimsen:
Well can you offer some
Hello,
Is there a possibility to connect from asterisk to mysql database without the
interface application like Ruby or PHP.
If i can connect to mysql database from asterisk, i can update the database for
manipulations.
Appreciate your response.
Regards,
Naveen.Palani
Try:
http://www.voip-info.org/wiki/view/Mysql
and the links thereon.
simon
Simon Elliston Ball
[EMAIL PROTECTED]
On 16 Jan 2008, at 19:11, Naveen Palani wrote:
Hello,
Is there a possibility to connect from asterisk to mysql database
without the interface application like Ruby or PHP.
I was pointed to the following:
http://asteriskforum.ru/viewtopic.php?t=1761
It is in Russian, which I don't speak, but it references an Asterisk patch.
Is this patch in 1.4.17?
Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?)
Anyone work with this?
Make sure asterisk is in the dialout group in /etc/passwd
The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, and
if you're using the gentoo ebuild of asterisk, it'll run as asterisk:asterisk,
so you need to make sure asterisk is a member of the dialout goup otherwise
On Wed, 16 Jan 2008 18:08:23 + (GMT), Gordon Henderson
[EMAIL PROTECTED] wrote:
However, you'll need to do similar things to your asterisk box router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the asterisk box,
On Wed, 16 Jan 2008 12:10:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
No, it cannot. You could use func_odbc to formulate your own queries,
though.
Thanks. I don't like ODBC, but if it's stable and not a pain to
install/use, that could be the solution.
Otherwise, there's a new solution
On Wednesday 16 January 2008 13:29:20 Simon Elliston Ball wrote:
Simon Elliston Ball
[EMAIL PROTECTED]
On 16 Jan 2008, at 19:11, Naveen Palani wrote:
Hello,
Is there a possibility to connect from asterisk to mysql database
without the interface application like Ruby or PHP.
If i
- Original Message -
From: Vincent [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 16 January 2008 22:01:55 o'clock (GMT) Europe/London
Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?
On Wed, 16 Jan 2008 18:08:23 + (GMT), Gordon Henderson
I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri
1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a
TE220B. The system load is below 0.10.
I moved the server into production, with one PRI, on Friday. On that
day we handled a couple thousand calls and
Steven wrote:
I'm not sure what to try next, other than calling the telco and asking
them to check their equipment. Does any one have a suggestion before I
do that?
I have a suggestion. Have you contacted Digium technical support for
assistance
with resolving this issue?
--
Russell
Hello users!
Recently I read that AgentCallbackLogin is going to be deprecated soon.
Wanting to set up a few callback type queues, I set them up as suggested
in queues-with-callback-members.txt.
I was able to set the queues up completely this way, however, I'm trying
to use Flash Operator Panel
On Jan 16, 2008 7:07 PM, Russell Bryant [EMAIL PROTECTED] wrote:
Steven wrote:
I'm not sure what to try next, other than calling the telco and asking
them to check their equipment. Does any one have a suggestion before I
do that?
I have a suggestion. Have you contacted Digium technical
Unbeatable price for a low end Asterisk server (or any server for that
matter)
http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=bednv4ks=bsd
I wonder if anyone has any experience with this box and Digium or Sangoma
hardware? Any compatibility issues? If not, I might
I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5
I having these problem :
Zap/2-1 is busy
Hangup ZAP/2-1
Everyone is busy/congested at this time (1:1/010)
Autofallthrough channel SIP/202-b7b08ab0 Status is busy.
And then HANGUP.
Trixbox 2.2... I assume you are using the latest version. Normally I
will ignore messages from trixbox users because they ask kindergarten
stuff... but you seem to be knowledgeable and I'll assume you chose
trixbox to make your life easier when it comes to dealing with others
regarding the PBX.
I
The phones are configured in the Users section of AsteriskGUI.
The bigger problem you'll have is that you probably also need to
replace/update the firmware on the 7910; by default they're configured to
work with Cisco's CallManager software. Start with this link:
I'm trying to test IMAP in 1.4.17 and it appears to be not working.
I've compiled imap-2007 with the following on a CentOS 5 box:
make slx EXTRACFLAGS=-I/usr/include/openssl -fPIC
and I've configured and compiled asterisk with the following:
./configure --with-imap=/usr/local/src/imap-2007
On Jan 16, 2008 6:39 PM, Steve Totaro [EMAIL PROTECTED] wrote:
Unbeatable price for a low end Asterisk server (or any server for that
matter)
http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=bednv4ks=bsd
I wonder if anyone has any experience with this box and Digium
On Jan 16, 2008 8:11 PM, Erik Anderson [EMAIL PROTECTED] wrote:
On Jan 16, 2008 6:39 PM, Steve Totaro [EMAIL PROTECTED]
wrote:
Unbeatable price for a low end Asterisk server (or any server for that
matter)
Has anyone tried installing Asterisk on ClarkConnect? It looks like
ClarkConnect runs on RHEL so it should work if they haven't modified it too
much.
It appears that ClarkConnect is working on adding Asterisk and integrating
it into their GUI but until then I'd also be interested in trying to
create nodes and links /proc/zap
On Jan 16, 2008 3:39 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
Make sure asterisk is in the dialout group in /etc/passwd
The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout,
and if you're using the gentoo ebuild of asterisk, it'll run
On Jan 16, 2008 7:28 PM, Steve Totaro [EMAIL PROTECTED] wrote:
You can add the raid option for $199. I think I might pickup about ten of
them at this price. I can always resell them as general purpose servers or
even workstations if Asterisk/Zaptel/Linux does not like the boxen.
Ahh - nice.
On Wednesday 16 January 2008 16:22:10 Vincent wrote:
On Wed, 16 Jan 2008 12:10:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
No, it cannot. You could use func_odbc to formulate your own queries,
though.
Thanks. I don't like ODBC, but if it's stable and not a pain to
install/use, that
any version of asterisk not create nodes into /proc/zap
create to command, view into make file how to create nodes
On Jan 16, 2008 8:48 PM, Walter Willis [EMAIL PROTECTED] wrote:
create nodes and links /proc/zap
On Jan 16, 2008 3:39 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
Make sure
And the problem is? ...
I think you should read this: http://catb.org/~esr/faqs/smart-questions.html
Regards,
Moisés Silva
On Jan 16, 2008 6:42 PM, Ruben Zamora [EMAIL PROTECTED] wrote:
I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5
I having these problem
The problem is that i have random hangup in calls in the PSTN.
After that I check in asterisk -rvv
Sip show channels
And I see the extension
The only way that I can place another call in the extension was to restart
the Asterisk.
-Mensaje original-
De: [EMAIL PROTECTED]
I'm trying to test IMAP in 1.4.17 and it appears to be not working.
I've compiled imap-2007 with the following on a CentOS 5 box:
make slx EXTRACFLAGS=-I/usr/include/openssl -fPIC
and I've configured and compiled asterisk with the following:
./configure
3Com http://www.sjc.cc.nm.us/documents/ots/docs/VoiceMailGuide.pdf
NEC Elitemail http://gigshowcase.com/EndUserFiles/2912.pdf
A system similar to Elitemail would rock!
Thanks,
Steve Totaro
On Jan 16, 2008 11:27 PM, Justin Newman [EMAIL PROTECTED] wrote:
Does anyone have flow charts or
I too would like this, Please feel free to post a link on the list :)
Regards
Kevin
Justin Newman wrote:
Does anyone have flow charts or digit/key cards for some of the more popular
voicemail systems out there?
(shows which digits/keys to press, where it takes you, etc.)
I need to create
I have the ones from T-Mobile Sprint PCS and probably the New ATT
Wireless... email me if you are interested.
On Jan 16, 2008 11:27 PM, Justin Newman [EMAIL PROTECTED] wrote:
Does anyone have flow charts or digit/key cards for some of the more popular
voicemail systems out there?
(shows
Oh and here's Audix: http://mx.netjdn.com/manuals/Legacy/quickref.pdf
Heck I'd think I'd put out a bounty on making Asterisk voicemail like
Audix if there were some people interested.
On Jan 16, 2008 11:27 PM, Justin Newman [EMAIL PROTECTED] wrote:
Does anyone have flow charts or digit/key
Hate to be spamming the list but I also have a list of every single
prompt in an Audix system if you want that I need to dig it up.
On Jan 17, 2008 12:05 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Oh and here's Audix: http://mx.netjdn.com/manuals/Legacy/quickref.pdf
Heck I'd think I'd put
Does anyone have flow charts or digit/key cards for some of the more popular
voicemail systems out there?
(shows which digits/keys to press, where it takes you, etc.)
I need to create some of the new voicemail system.
Send 'em my way if you have them.
nt_jnewman at yahoo.com
Justin
hi all,
how to set the caller id facility for
the TDM400p card.
Please help me
thanks,
sandeep.s
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 15, 2008 3:09 PM
Subject: asterisk-users Digest, Vol 42, Issue 51
Send
Hello!
Robert Moskowitz wrote:
I was pointed to the following:
http://asteriskforum.ru/viewtopic.php?t=1761
It is in Russian, which I don't speak, but it references an Asterisk patch.
Is this patch in 1.4.17?
Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?)
Anyone work
We've implemented the Audix-similar system. I'd welcome a bounty! ;)
Please send over cards for the other two systems you mentioned...
- Original Message
From: Andrew Joakimsen [EMAIL PROTECTED]
To: Justin Newman [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Let me see how much work it would be...might be able to include those as well.
- Original Message
From: Steve Totaro [EMAIL PROTECTED]
To: Justin Newman [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday, January
Hi there,
We are looking for a CompactPCI (cPCI) blade that supports 2 or more FXO
modules and is operable with Asterisk PBX.
We have some adlink 6U cPCI servers running Asterisk perfectly as a SIP only
solution and have some other PC based boards handling the zaptel FXO cards
but it would be
I am trying to configure Asterisk for BSNL, india network.
I have successfully configured it for outgoing calls.
When any outside number make any call to trunk then it receives the call
properly but when the call is disconnected by inside extension then outside
phone does not get a busy tone.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Justin Newman wrote:
Let me see how much work it would be...might be able to include those as well.
I'm assuming you've also seen Jeffrey C Ollie's Asterisk Voicemail User
Reference?
http://www.venturevoip.com/news.php?rssid=388
- --
Kind Regards,
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