On Wed, Apr 22, 2009 at 11:05:38AM +0530, Kurian Thayil wrote:
> On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
> > Daily Asterisk restart
>
> Do you think its mandatory in production env?
No.
>
> >
> > Daily log rotation
A simple logrotate file takes care of that.
--
I saw a few posts of this problem before I could not figure out the
reason I am getting it.
I am running RHEL 5, Asterisk 1.4.21.2, zaptel 1.4.11 and libpri 1.4.4
Basically, if I dial into a queue and hang up the phone, Asterisk did
not detect the hangup request and Asterisk will only hang up whe
2009/4/22 michel freiha
> Hi all,
> Does asterisk support the following scenario? I need when a customer who
> own an endpoint registered on asterisk open the line, the asterisk will run
> a specific AGI script inside the endpoint context?
>
You mean when they pick up the phone it'll automatical
> > Daily Asterisk restart
>
> Do you think its mandatory in production env?
>
It could be a pre-1.6 advice but I still stick to it.
I did it to all my production Asterisk servers.
___
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2009/4/22 Michael
> I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing.
>
> Does anyone know of a way, either while producing the file, or after, to
> tell
> how many pages have been produced? (without manually viewing the file)
>
tiffinfo? then count the number of data blocks...
On Wed, 22 Apr 2009, James Mutuku wrote:
> I know this might be test book question or one best suited for google
> but I will take the risk of asking. Here I go. What common routine
> maintenance tasks do you run on your asterisk box?
None.
I configure Asterisk to log everything to syslog on a
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
> Daily Asterisk restart
Do you think its mandatory in production env?
>
> Daily log rotation
>
> Voicemail clean up for people leaving an organization.
>
>
>
>
> ___
Daily Asterisk restart
Daily log rotation
Voicemail clean up for people leaving an organization.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Mutuku
Sent: Wednesday, 22 April 2009 3:15 PM
To
Hello(s),
I know this might be test book question or one best suited for google but I
will take the risk of asking. Here I go. What common
routine maintenance tasks do you run on your asterisk box?
Thanks
James.
___
-- Bandwidth and Colocation Provided b
I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing.
Does anyone know of a way, either while producing the file, or after, to tell
how many pages have been produced? (without manually viewing the file)
Michael
___
-- Bandwidth and Colocat
"
Hello,
If anybody has any idea's to where I should start looking to fix the below
subscription problem. If there is another mailing list I should post this to
please let me know.
Thank you,
Brad Finberg
- Original Message -
From: Brad Finberg
To: Asterisk Users Mailing List - Non-
The Asterisk Development Team is pleased to announce the fifth release
candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc5 is available for
immediate download at http://downloads.digium.com/pub/asterisk/
This release fixes a couple of issues with realtime music on hold that could
cause Asterisk to
In this case I had, in my hurry this morning, simply confused G.722.1C
and G.722.2. These are both low bitrate wide bandwidth codecs.
They are also known by the Polycom marketechure nomenclature of Siren7
and Siren14. G.722.1 supporting 7 KHz passband, while G.722.1C support
14 KHz passband.
Mic
On Tue, Apr 21, 2009 at 4:40 PM, Steve Underwood wrote:
> Which Polycom supports G.722.2? I think they are only supporting G.722,
> G.722.1 and G.722.1C right now.
Could someone enlighten me, what is the difference (the result part
that matters, not the spec)?
r
Even if Zaptel is compiled, you can also compile Dahdi because Asterisk
will choose the DAHDI-module... it seems.
So I left Zaptel... and compiled Dahdi (everything went well, I followed
the steps) en then Asterisk again (with dahdi support!!).
Yet another episode in this nightmare :
[r...@aster
Benny Amorsen writes:
> Asterisk DB is either an SQLite database or a Berkeley database, I
> forget which (did it change?). Either way, 20,000 should be a problem
> for the underlying database.
Should NOT be a problem for the underlying database.
Sorry!
/Benny
__
On Tue, 21 Apr 2009, Tzafrir Cohen wrote:
> WAV is a pretty simple container format. The length is written in a very
> expected place in the header:
>
> http://en.wikipedia.org/wiki/.wav
> http://ccrma.stanford.edu/courses/422/projects/WaveFormat/
>
> E.g. the following:
>
> wav_size() {
>
On Fri, Apr 17, 2009 at 11:39 AM, Mike wrote:
> True, my mistake: I upgraded to 1.4.24.1, and the MoH file still always
> start from the beginning.
I believe I'm experiencing the same thing with my music on hold. I
also would prefer a continuous play in the background, and I'm using
asterisk-1.6.
Hi all,
Does asterisk support the following scenario? I need when a customer who own
an endpoint registered on asterisk open the line, the asterisk will run a
specific AGI script inside the endpoint context?
Regards
___
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On 21-Apr-09, at 1:06 PM, Anthony Francis wrote:
>> You are correct, not seeing that means that the signaling was
>> either in
>> the audio stream (which doesn't survive compression) or it was sent
>> in
>> the sip signaling. However one must also note that your ITSP's
>> gateway
>> may have
On Tue, 21 Apr 2009, Doug Lytle wrote:
> Benny Amorsen wrote:
>> Asterisk DB is either an SQLite database or a Berkeley database, I
>>
>
> The last I knew, it was BerkeleyDB.
>
> Doug
>
>
Just to add a few cents, if the object is to store and retrieve a single
value with a single key, Berkeley
On Tue, 21 Apr 2009, Benny Amorsen wrote:
> "Sriram" writes:
>
>> 1. I need to store the CallerId of the PSTN caller with his language
>> preference so that next time he is played the prompt in his language that
>> he chose the first time.What would be better - storing his number in the
>> Asteri
Anthony Francis wrote:
> Jeff LaCoursiere wrote:
>
>> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>>
>>
>>
>>> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>>>
>>>
>>>
> I went ahead and switched to SIP just for grins, and made sure
> "dtmfmode=rfc2833" is in the pee
Benny Amorsen wrote:
> Asterisk DB is either an SQLite database or a Berkeley database, I
>
The last I knew, it was BerkeleyDB.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
__
Jeff LaCoursiere wrote:
> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>
>
>> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>>
>>
I went ahead and switched to SIP just for grins, and made sure
"dtmfmode=rfc2833" is in the peer config on both sides and in the entry
for the phon
I second the "Real" database idea. AFAIK, the Asterisk database is still a
Berkley DB. I'm accessing Postgres using an AGI and returning dialplan
variables with what I want to process. The Asterisk database is best for
small, non-critical information, though there are good procedures documented
"Sriram" writes:
> 1. I need to store the CallerId of the PSTN caller with his language
> preference so that next time he is played the prompt in his language that
> he chose the first time.What would be better - storing his number in the
> Asterisk DB and using Dbput and DBget ? or storing it in
Hi All,
Thanks for your answers.
Asterisk (v1.2.7.1) runs on RedHat AS 4 without -g option.
It's really a crash, the process not running at all according to "ps aux".
Regards,
A.L
___
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On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>
> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>
>>> I went ahead and switched to SIP just for grins, and made sure
>>> "dtmfmode=rfc2833" is in the peer config on both sides and in the entry
>>> for the phone. So now it is:
>>>
>>> polycom501---SI
Marco Sambo escribió:
Hi,
I have the same problem: sometimes my Asterisk box crash (or similar)
and in asterisk log doesn't appear nothing. Also into syslog.
I don't understand what is it
Marco
Well, when asterisk dies "without leaving trace", it's generally a core
dump. That means at as
mgra...@mstvp.com wrote:
> Doing a little research before Friday's Voip Users Conference call with
> Dan Behringer.
>
> Are any of the newer Polycom wideband codecs implemented in v1.6?
> Specifically, G.722.1 or G.722.2?
>
Which Polycom supports G.722.2? I think they are only supporting G.722,
Is it a crash or you find that the phones are not registered ? Do you loose
internet connection during the night ? Do you have SIP trunks ? Is it
vanilla asterisk or a specific distro ?
G.
On Tue, Apr 21, 2009 at 6:25 AM, Barry L. Kline wrote:
> Adrien Lemoine wrote:
>
> > Maybe someone experienc
mgra...@mstvp.com wrote:
> Doing a little research before Friday's Voip Users Conference call with
> Dan Behringer.
>
> Are any of the newer Polycom wideband codecs implemented in v1.6?
> Specifically, G.722.1 or G.722.2?
Asterisk 1.6 has passthrough/record/playback support for G.722.1
(Siren7) a
i'd use mysql... and i do use mysql for this...
2009/4/21 Sriram
> My setup : Trixbox 2.6.1 & TE410P running well .:
>
> 1. I need to store the CallerId of the PSTN caller with his language
> preference so that next time he is played the prompt in his language that he
> chose the first time.Wha
Doing a little research before Friday's Voip Users Conference call with
Dan Behringer.
Are any of the newer Polycom wideband codecs implemented in v1.6?
Specifically, G.722.1 or G.722.2?
Thanks,
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype
On Mon, Apr 20, 2009 at 10:45:00AM -0400, Justin Piszcz wrote:
> Hello,
>
> When a voice message is saved and e-mailed as a wav, the total time of the
> voice mail does not show up in, e.g., windows media player, why is this?
>
> I have only used wav49/wav:
>
> ; Use wav49 format for all voicem
Adrien Lemoine wrote:
> Maybe someone experienced something similar and can drive me in the
> resolution ?
You have given no information about your hardware, OS, Asterisk version
or what you need to do to recover the system (e.g. reboot, just restart
Asterisk, etc) so no one is going to be able t
On Tuesday 21 April 2009 08:07:17 Justin Piszcz wrote:
> On Mon, 20 Apr 2009, Justin Piszcz wrote:
> > When a voice message is saved and e-mailed as a wav, the total time of
> > the voice mail does not show up in, e.g., windows media player, why is
> > this?
> >
> > I have only used wav49/wav:
> >
Hi,
I have the same problem: sometimes my Asterisk box crash (or similar) and in
asterisk log doesn't appear nothing. Also into syslog.
I don't understand what is it
Marco
2009/4/21 Adrien Lemoine
> Hi all,
>
>
>
> I experienced for a second time the crash of asterisk process during th
--[ UxBoD ]-- wrote:
> I am going to be building a new home Asterisk server this weekend
> (Dual core Intel Atom & 2GB RAM) and would like to ask whether it
> would be worth starting fresh with a 1.6 install instead of the 1.4
> one I have at the moment ? I do not have a complicated dialplan as it
On Mon, 20 Apr 2009, Justin Piszcz wrote:
> Hello,
>
> When a voice message is saved and e-mailed as a wav, the total time of the
> voice mail does not show up in, e.g., windows media player, why is this?
>
> I have only used wav49/wav:
>
> ; Use wav49 format for all voicemail messages
> format
Hi,
I am going to be building a new home Asterisk server this weekend (Dual core
Intel Atom & 2GB RAM) and would like to ask whether it would be worth starting
fresh with a 1.6 install instead of the 1.4 one I have at the moment ? I do not
have a complicated dialplan as it only serves a couple
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Sriram wrote:
>
> 1. I need to store the CallerId of the PSTN caller with his language
> preference so that next time he is played the prompt in his language
> that he chose the first time.What would be better - storing his number
> in the Asterisk DB
My setup : Trixbox 2.6.1 & TE410P running well .:
1. I need to store the CallerId of the PSTN caller with his language preference
so that next time he is played the prompt in his language that he chose the
first time.What would be better - storing his number in the Asterisk DB and
using Dbput a
I am using IBM Server I cleared the event log from BIOS
and asterisk couldn't run
which file i have to create ?
and what is its permission?
thanks a lot
_
Show them the way! Add maps and directions to your party invites.
http://www
Hi all,
I experienced for a second time the crash of asterisk process during the
night.
Nothing in Asterisk messages logs, nothing in /var/log/messages can explain
that.
Maybe someone experienced something similar and can drive me in the
resolution ?
Regards,
A.L
__
On 21 Apr 2009, at 10:46, Khaled W. Chehab wrote:
> Dears,
Hi..
> When my GW send a call to asterisk v 1.4.24 ,
What is your GW. Hardware, software etc etc
> Asterisk send Status: 420 bad extension (unsupported)
Ok. SIP trace available?
> Why?
Show us the logs/sip trace.
> Any modi
Dears,
When my GW send a call to asterisk v 1.4.24 ,
Asterisk send Status: 420 bad extension (unsupported)
Why? Any modifications should be done one sip.conf
regards
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