On Thu, 7 Jun 2007, Stewart Nelson wrote:
You can reload via http using a command like:
wget\
--output-document=/dev/null\
--quiet\
http://ip-address-of-pap/upgrade?http://ip-address-of-web-
server:80/asterisk/spa000F66A83C90.cfg
I tried it with my xml
Hi guys,
I was wondering whether there's anyone who could share his/her
experiences with using Microsoft RTC Library. In particular I am
wondering what Ethernet capacity should I have in scenario of 30 people
using Microsoft RTC Library for SIP communication (PBX is obviously
Asterisk :-) )
The setup.
Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
updates applied), the TE410 lives on it's own interrupt.
Asterisk sits between our telco and a PRI enabled PBX.
These are the relevant versions installed:
Linux: 2.6.20-1.2316.fc5smp
Zaptel: 1:1.4.2.1-34.fc5
Asterisk:
Nick,
Pretty much - it builds the XML output on the fly, and delivers it
over HTTP (PHP/Apache).
It works the same for all of the mainstream Linksys kit - including
SPA phones.
Generally, where we've installed IP phones, we've also installed an
Asterisk appliance in the form of a Linux
Hi,
I have Asterisk 1.4.4 on my linux box.
Whenever i try to kick a participant in conference say 59681446 using
following command
meetme kick 59681446 1
where 1 is the participant number, following are the actions that
asterisk takes
* IVR You have been kicked from this conference is
On Fri, 8 Jun 2007, Asterisk wrote:
Hi guys,
I was wondering whether there's anyone who could share his/her
experiences with using Microsoft RTC Library. In particular I am
wondering what Ethernet capacity should I have in scenario of 30 people
using Microsoft RTC Library for SIP communication
On Thu, Jun 07, 2007 at 04:50:55PM -0600, Stephen Bosch wrote:
Tzafrir Cohen wrote:
To generate a FXS dialtone without Asterisk, use fxstest (make fxstest)
from the zaptel source directory.
Can I break this dial tone with DTMF?
No.
--
Tzafrir Cohen
icq#16849755
I'm talking out my rear so someone please apply an attitude
adjustment if I'm way off base.
But, if you are using Dundi as a lookup engine it should know the
contact information both endpoints and how to reach them perhaps not
ONLY knowing how to comunicate via another asterisk box.
Much
In this troubleshooting case, it probably is better that there is NO
dialtone, which would make the hiss easier to hear.
I am curious what the OP found
When Asterisk is stopped, does the hiss continue?
That would help to narrow down the location of the problem
It sure sounds to me as if it is a
Hello Matthew,
Java is not a great solution for AGIs because they are script you should
fire up and terminate very fast, while the overhead of launching a JVM,
loading all classes, etc, is pretty large. Also, you don't want multiple
JVMs in parallel loading everything multiple times.
John Novack wrote:
In this troubleshooting case, it probably is better that there is NO
dialtone, which would make the hiss easier to hear.
I am curious what the OP found
When Asterisk is stopped, does the hiss continue?
That's tough to assess because the other problem I have had with it is
-163c, record-enable|2000|IN) in new stack
-- Executing GotoIf(SIP/4000-163c, 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI(SIP/4000-163c,
recordingcheck|20070608-131412|1181308451.0) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Did it accompany an update you made? If you can find out what version
the problem started occurring, that would help in fixing the problem.
Matthew Fredrickson
Software/Hardware Engineer
Digium, Inc.
On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote:
The setup.
Asterisk is on a 3G Zeon
Hi,
I have my custom sounds which should be played instead of the default
ones when a user is busy or unavailable:
The person at extension XXX is not available right now, please
Of course I can simply replace the files, but the problem is my
implementation shouldn't (MUST NOT) mention the
On 6/8/07, Matthew Pease [EMAIL PROTECTED] wrote:
when will it be out?
Soon... it's going through the copyediting process right now. I can't
give any more specific timeframe than that, as I don't know how long
it'll take to get through the entire process, but if I had to make a
wild guess I'd
On Fri, Jun 08, 2007 at 02:52:39PM +0100, Carlos Jerónimo wrote:
Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk.
I've sucessfully installed it with the command:
#apt-get install asterisk
Then after installing FreePBX i get this error when restarting asterisk:
[EMAIL
Dovid,
Please provide a simple network diagram for members of this list. Q931
cause 44 error is a layer 3 ISDN error (Requested circuit/channel not
available) most likely mapped backwards from PRI T1 interworking.
John Treble
Ottawa, Canada
From:
Having a problem w/ not getting CID name from a PRI. CID Name appears in
the PRI debug, but even after a Wait(4) it still appears after the phone is
ringing. Here is the relevant info from my PRI debug output. Line 4 is a
NoOp showing me trying to echo Name and Number. Line 6 dials the
On Fri, 8 Jun 2007, Jared Smith wrote:
On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote:
Still need an answer to this one.
I wrote a response yesterday, but it looks like it didn't come through
for some reason. The answer is to use the DumpChan() application and
watch the CLI when it's
This is off topic for Asterisk but I need a suggestion. I have a
customer (travel agency) that has recently begun complaining that their
GXP-2000 phones are getting very hot, they say that around mid day the
handset is so hot that it can burn your ear. These phones are in
constant use
Timothy Parez wrote:
Hi,
I have my custom sounds which should be played instead of the default
ones when a user is busy or unavailable:
The person at extension XXX is not available right now, please
Of course I can simply replace the files, but the problem is my
implementation shouldn't
I know that g729 is the king-all, but I want to know what the rest of
the professional are using out there. g729 has a cost involved, so does
the cost really offset the performance? Or is it better to go with g711
to start off?
I'm wary of using g711 of public broadband networks. Although
Hey guys,
I'm currently running Asterisk 1.2.18 Under Mandriva Linux. Three
Facilities are hooked together via IAX2 (Trunked) over a OpenVPN
connection on a 10mbit (uplink/downlink) internet connection. I was
parked for around thirty seconds at a remote facility. All of a sudden,
the call
Barton Fisher wrote:
Anybody have an answer? TIA
This is really strange. Every message to the (VGA) console is written
twice to the screen, but not on the SSH connection.
Any clues how to stop this behavior?
-- Executing BackGround(Zap/216-1, custom/3566/91_|m|) in
new stack
--
Yes. In fact it's around 32kbps, for a high duration call. MRTG statistics.
Are you using G729A ou B? (VAD can reduce the usage).
Att, Ricardo Martins.
Henry Cobb escreveu:
On 6/7/07, Ricardo Martins [EMAIL PROTECTED] wrote:
We use G.729. Consumes only 35kbps of bandwidth and has a level 4
Lenz wrote:
Hello Matthew,
Java is not a great solution for AGIs because they are script you should
fire up and terminate very fast, while the overhead of launching a JVM,
loading all classes, etc, is pretty large. Also, you don't want multiple
JVMs in parallel loading everything multiple
It probably did but we run in updates every week and nobody can state
exactly when the problem started only a few weeks ago - not very
helpful.
I can see that when I hear the issue the iowait time is high on the
processor.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello,
I have an asterisk 1.2.18 working fine, the only problem is that all
applications that play audio, sound like tremolo or vibrato, but
musiconhold plays fine.
The same audio file (wav, mp3, ...) works fine with Musiconhold()
but not with Playback() or Background()...
Carlos,
We had this happen once here with a batch of phones received from
Grandstream about a year ago now. Email Grandstream on it and they
should know exactly what the problem is, I believe they ended up
replacing the phones for us.
Jessee Holmes
Atacomm / Ataractic Corporation
I'm wary of using g711 of public broadband networks.
...
It'd be interesting to see some comparisons or comments from
people using g726 as this does seem to be supported by quite
a few hardware devices.
We are using g711 pretty much exclusively for all residential
customers in the US and it
Hi,
Can Asterisk write to multiple MySQL databases in different machines,
at the same time, as a backup scheme?
If it does, where can that be configured? In res_mysql.conf file?
Does anyone ever made it?
Regards,
Ricardo.
___
--Bandwidth and
Lee Jenkins wrote:
sathish s wrote:
i need to catch the call hold event from my asterisk-java program.
Im using net.sf.asterisk.*; for communicating with asterisk server.
I need to get the call hold status on my java program . I can able
to get the music on hold status but i cannot able to
On 6/8/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Log into your mailbox. Press 0, then press the option listed to
record your unavail and busy greetings.
I'm no expert, so someone feel free to correct me if I'm wrong, but
you should be able to make one or two recordings and then
On 6/8/07, Kyle Sexton [EMAIL PROTECTED] wrote:
Having a problem w/ not getting CID name from a PRI. CID Name appears in
the PRI debug, but even after a Wait(4) it still appears after the phone is
ringing. Here is the relevant info from my PRI debug output. Line 4 is a
NoOp showing me trying
MySQL has its own ways of doing this kind of thing. Take a look at the
documentation http://dev.mysql.com/doc/refman/5.0/en/replication.html on
MySQL's website related to replication.
Bobby
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent:
It would be better to let MySQL handle that - use the built-in
replication facilities. It's easy to setup.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been
Why would you do this why put the overhead inside asterisk when
mysql has perfectly good replication mechanisms built in?
On Jun 8, 2007, at 12:44 PM, [EMAIL PROTECTED] wrote:
Hi,
Can Asterisk write to multiple MySQL databases in different
machines, at the same time, as a backup
[EMAIL PROTECTED] wrote:
Hi,
Can Asterisk write to multiple MySQL databases in different machines,
at the same time, as a backup scheme?
If it does, where can that be configured? In res_mysql.conf file?
Not that I'm aware of, but you can setup MySQL to mirror the data to a
slave database.
Justin Moore wrote:
On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
It would be better to let MySQL handle that - use the built-in
replication facilities. It's easy to setup.
That's a great idea for backup purposes, but if the OP is wanting true
redundancy, that won't help much. What
Java is not a great solution for AGIs because they are script you should
fire up and terminate very fast, while the overhead of launching a JVM,
loading all classes, etc, is pretty large. Also, you don't want multiple
JVMs in parallel loading everything multiple times.
How about writing your
Eric ManxPower Wieling wrote:
This is really strange. Every message to the (VGA) console is
written twice to the screen, but not on the SSH connection.
Any clues how to stop this behavior?
Stop running in graphics mode.
OK, that's a great clue, but can you tell me how to disable now?
On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
It would be better to let MySQL handle that - use the built-in
replication facilities. It's easy to setup.
That's a great idea for backup purposes, but if the OP is wanting true
redundancy, that won't help much. What happens then when the
On Thu, Jun 07, 2007 at 04:52:31PM -0400, Jared Smith wrote:
On 6/7/07, Matthew Pease [EMAIL PROTECTED] wrote:
I'm having awesome fun with Asterisk voicepulse connect together.
So cool.
I'm glad you're having fun!
I'm trying to have the caller id read back to me.Do I need to
iowait time? I'm not familiar with that. Where are you seeing that?
Also, is it a reproducible problem?
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote:
It probably did but we run in updates every week and nobody can state
exactly
Timothy Parez wrote:
I have my custom sounds which should be played instead of the default
ones when a user is busy or unavailable:
The person at extension XXX is not available right now, please
Of course I can simply replace the files, but the problem is my
implementation shouldn't
Hi,
One of my users is in trouble with his polycom phone hooked to an
asterisk server.
The phone works fine for a few days, and then disappears from the
registered sip peers in asterisk.
The user is able to place outbound phone calls, but can't receive
incoming calls until the network plug is
Hi,
I need to build Text entry application by using asterisk. I already tried
this with spandsp application along with app_dtmftotext.c file, it was not
working because of some version problem.
Is there any way of building the text entry application through touch pad.
Regards
K.Rajesh.
On Fri, 2007-06-08 at 11:12 -0600, Anthony Francis wrote:
Lee Jenkins wrote:
sathish s wrote:
i need to catch the call hold event from my asterisk-java program.
Im using net.sf.asterisk.*; for communicating with asterisk server.
I need to get the call hold status on my java program .
I am trying to set up somthing so I can dial into my asterisk box, and
have it behave as a modem bank. Is there anything like that already, or
am I going to have to write my own. I checked googls and found no
leads, but thought I would ask here before I tried writing my own, just
to make
UltraMonkey (www.ultramonkey.com) and MySQL Cluster
(http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster.html)
It works a charm.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Justin Moore
Sent: Friday, June 08, 2007 2:13 PM
To: [EMAIL
Hi,
Are there any decent (commercial or free) LOG parsers for A*k.
Its *really* hard to debug issues involving multiple calls (eg meetme)
when all of the messages are interlaced with each other. There must be
an easier way. (A*K 1.2.18)
Adrian
___
On 6/7/07, JR Richardson [EMAIL PROTECTED] wrote:
Hi All,
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
Under example:
exten = s,2,Set(CDR(MyFavoriteBand)=Foo Fighters)
exten = s,3,Set(CDR(MyFavoriteSong)=Hero)
and under description:
-userfield: The channel's user specified
Can Asterisk write to multiple MySQL databases in different machines,
at the same time, as a backup scheme?
If it does, where can that be configured? In res_mysql.conf file?
No, you cannot write to 2 different mysql servers with res_mysql.
Just use MySQL replication as an alternative. Easy to
On 6/8/07, Justin Moore [EMAIL PROTECTED] wrote:
On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
It would be better to let MySQL handle that - use the built-in
replication facilities. It's easy to setup.
That's a great idea for backup purposes, but if the OP is wanting true
Hi, tahnks for your answer. but i haved install with command apt-get
install asterisk, but i don't have package asterisk-addons.
if i download asterisk-addons by digium site, run well with asterisk
debian pakages??
thanks
2007/6/8, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Jun 08, 2007 at
We have a customer with an obsolete Centigram voicemail system who would
like to replace it with Asterisk.
Any one with experience doing this or information on the signalling and
trunking used to connect the Mitel SX-2000 to the Centigram server?
--
George Pajari (dCAP), netVOICE
On 6/8/07, Christopher Dobbs [EMAIL PROTECTED] wrote:
I am trying to set up somthing so I can dial into my asterisk box, and
have it behave as a modem bank. Is there anything like that already, or
am I going to have to write my own. I checked googls and found no
leads, but thought I would ask
The IAXMODEM might get you half way there...but if you want to connected it
to a windows box (which I assume is why you use the RAS acronym), you'll
have to look for remote serial port software.
-MD-
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Fri, Jun 08, 2007 at 08:24:45PM +0100, Carlos Jerónimo wrote:
Hi, tahnks for your answer. but i haved install with command apt-get
install asterisk, but i don't have package asterisk-addons.
if i download asterisk-addons by digium site, run well with asterisk
debian pakages??
Not exactly.
On Fri, 8 Jun 2007, Paco Brufal wrote:
Hello,
I have an asterisk 1.2.18 working fine, the only problem is that all
applications that play audio, sound like tremolo or vibrato, but
musiconhold plays fine.
The same audio file (wav, mp3, ...) works fine with Musiconhold()
but not
Hi,
We have a PRI connection when its was on test networks we had echo
problems withoutside line.
So I bought a TE212P card resolve the echo problem. Which did to an extent.
Its using asterisk 1.2.18 RHEL4-Update 4.
But now when we are live, there is a terrible echo between 2
Moy:
I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem
i have is the RxFAX application, that broke every time... With and error in
the linking to the spandsp library.
If i have time this weekend i will review to fix the app,
Thanks.
On 6/4/07, Tobias Wolf [EMAIL
On Sat, 9 Jun 2007, Deepak Naidu wrote:
But now when we are live, there is a terrible echo between 2 SIP calls.
If I call the same extension from outside the voice is clear.
My impression is that the transcoding that takes place between two
purely software SIP calls never goes through the
[EMAIL PROTECTED], you are email bombing me, please fix your blackberry!
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, June 08, 2007 7:38 PM
To: Steve Totaro
Subject: Delivery Status
I have a small call center running with Asterisk 1.4.4 and Unicall.
Everything seems to be working but twice now we had to reset the server
because all lines stopped working. You can see users dialing in and
reaching the queue but the agents never get the call and the lines are
not
Thnx for your quick replies.
I will try all of the above methods :-)
On Fri, 2007-06-08 at 13:02 -0400, Justin Moore wrote:
On 6/8/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Log into your mailbox. Press 0, then press the option listed to
record your unavail and busy greetings.
Hi!
Hope someone can help me. I'm trying to pass SIP traffic from one asterisk
to another through a third server. Here is the desired scenario:
ServerA -- SIP -- ServerB -- SIP -- ServerC
When a call is placed on a ServerA local, I can see that ServerB receives
the call and dials ServerC. But
Ya, I have done that, below is zapata.conf. Also we had an TMP card with
analog lines. SIP cals were great on them. now when we switched over. SIP
calls have echo.. which shouldnt be at all.
[channels]
language=en
#include zapata_additional.conf
context=from-pstn
switchtype=national
Hi all users,
I has been joining this user list for about 1 year, and always has seen the
successful story about the Asterisk act as IP PBX and even communication
appliances solutions. And thank for this list to help each other and make
everyone success. I also being inspired by this user-list
looking for good sip softphone for wifi and 3G network.
1 are there any sip softphone ( with gsm/g723/G729 codec ) for
smartphone such as Nokia N90 / 93 / N95 ?
2 are there any sip softphone ( with gsm/g723/G729 codec ) for Window
mobile5 Or wm2003 ?
3 How is the sound
On Fri, 8 Jun 2007, Jared Smith wrote:
On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote:
Still need an answer to this one.
I wrote a response yesterday, but it looks like it didn't come through
for some reason. The answer is to use the DumpChan() application and
watch the CLI when it's
On Fri, 8 Jun 2007, Steve Edwards wrote:
On Fri, 8 Jun 2007, Jared Smith wrote:
On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote:
Still need an answer to this one.
I wrote a response yesterday, but it looks like it didn't come through
for some reason. The answer is to use the
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