in a simple dialplan like follows:
[firstcontext]
include = secondcontext
include = thirdcontext
include = fourthcontext
[fourthcontext]
_03X.,1,Goto(${EXTEN:2},1)
_X.,1,DoSomething()
_X.,2,Hangup()
the Goto() for exten _03X. seems to start the search for the jump within
firstcontext, thus
Already using the CDR(userfield), and overloading it with multiple variables
will make some DB operations nastier to work with (I'm a little fuzzy and vague
on exactly what... something to do with sql joins?). I'm digging deeper into
how much pain it really causes us on the DB/App side to see
Hi all,
I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I
make a call I get this:
Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception
on 19, channel 1
Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627
Still i cannot resolve this issue, please anyone can help me with this?
Thanks in advance
--
_
Facundo Agustin Barrera
--
www.openlabs.com.ar
Let the penguins do the work
-
Buenos
On Mon, 8 Jan 2007, Rajkumar S wrote:
Hi,
This is slightly off topic, but here I go any way...
VoIP traffic has lot's of smaller packets, and since each packet can
generate an interrupt, is there any way to determine the irq rates in
a machine, and more importantly to know if I am hitting any
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Yuan LIU wrote:
One item in my todo list is to make better sound quality whenever end
point supports it. Wide-band codec's can already produce better sound
than toll. So why do we still need to convert to 8 bit?
Should be 16 bit, 8Khz, not 8
Hi,
I need some help on how to manage the full log file. It's getting
quite large now and I'd like to clear it. Is there any simple command
for this or should I just delete the file (need to be sure this won't
affect the system).
Also - how do I keep the log file from growing so large?
Thanks!
/var/lib/asterisk/licenses
:-)
On 1/8/07, Xue Liangliang [EMAIL PROTECTED] wrote:
Hi, leo, I will try the following solution that seperate
/usr/lib/asterisk/modules in another patition other than drbd, then
register the licenses on both server. not sure where the license key
acutally lies
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI logger reload command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
Thanks for the quick response!
I read about logrotate at voip-info.org but I didn't quite understand it. I'm
no asterisk/linux expert unfortunately.
First of all. What exactly does happen when I run:
/usr/sbin/asterisk -rx 'logger rotate'
Does it clear the file and create a new one? Can I run
Yes, trixbox includes more bloat and more problems. Let's clarify
something, trixbox includes FOP which is part of FreePBX!
-Original Message-
From: Steve Sobol [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 07, 2007 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Mon, Jan 08, 2007 at 04:40:14AM -0800, shadowym wrote:
Yes, trixbox includes more bloat and more problems. Let's clarify
something, trixbox includes FOP which is part of FreePBX!
FOP is actually an independent application.
--
Tzafrir Cohen
icq#16849755
but keep in mind, that jb for sip (generic jitterbuffer) is implemented
differently, than iax, so it works only for SIP-SIP calls, or SIP-ZAP
and, curious, eg. for SIP-ZAP call must be activated for (outgoing) ZAP
channel :-\
yusuf wrote:
[EMAIL PROTECTED] wrote:
In iax.conf there is
[EMAIL PROTECTED] schrieb:
Thanks for the quick response!
I read about logrotate at voip-info.org but I didn't quite understand it. I'm
no asterisk/linux expert unfortunately.
First of all. What exactly does happen when I run:
/usr/sbin/asterisk -rx 'logger rotate'
Does it clear the file and
Hi Jan,
You should use the logrotate in order to delete the log on periodic
intervals. This article is meant to do exactly the opposite :)
http://astrecipes.net/index.php?n=205 but you get an idea of how to setup
log file rotation and how to notify Asterisk that it should open a new
file
hi list,
after connecting 3 asterisk servers via IAX in a line
(+ 1 client at each end), i noticed that call path
optimization happens only one time, i.e. only one
node/leg in the path can be reduced.
Does anybody know if this is the intended behaviour or
if it's a bug?
Can anyone confirm my
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Julian Lyndon-Smith wrote:
Is there anyone with any experience of using the AMD app and the
settings that worked for them in the UK ?
Any help would be appreciated.
Hi, I'm using it in New York, and we seem to be having good success (on
this
Super! Thanks! Now I see how the script works a bit more clearly. :)
I still don't understand what happens if I run:
/usr/sbin/asterisk -rx 'logger rotate'
Can I run the above without having the script? What will the command do?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL
Hi Yusuf, how are you?
It orders in the list its configurations, so that let us can help.
Best Regards
Josue
2007/1/8, yusuf [EMAIL PROTECTED]:
Hi all,
I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a
Sangoma A101, and when I
make a call I get this:
Jan 8 13:04:06
On Mon, Jan 08, 2007 at 02:28:19PM +0100, [EMAIL PROTECTED] wrote:
Super! Thanks! Now I see how the script works a bit more clearly. :)
I still don't understand what happens if I run:
/usr/sbin/asterisk -rx 'logger rotate'
Can I run the above without having the script? What will the
great help. Thanks for that Matt.
One thing that is really confusing me at this point: if I want to leave
an automated answer machine message, and amd tells me it's a machine,
how do I know when to start leaving the message ? Some intros are long
(thanks for calling, me and mine are not here
I am an addict to teasin. Takes one to know on ;)
- Original Message -
From: Tom Lynn
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, January 01, 2007 7:12 PM
Subject: Re: RE : [asterisk-users] Happy 2007!!!
Dovid, you're killing me. This after
I connect to a PSTN carrier over SIP which requires me to connect with
a g729 codec. I'm using them for just robocalling: Asterisk server
originates calls which play a prerecorded file. Can I pre-encode those
stored files in g729 so they don't need to be encoded for each call? If
so, do I
Matthew Rubenstein wrote:
I connect to a PSTN carrier over SIP which requires me to connect with
a g729 codec. I'm using them for just robocalling: Asterisk server
originates calls which play a prerecorded file. Can I pre-encode those
stored files in g729 so they don't need to be encoded
Thank you, that is excellent advice.
I understand that Intel has a free g729 codec, and that there might be
others. Free g729 codecs cheat Digium of some income that helps keep
them producing free Asterisk (and hosting lists like this one), but what
other reasons (quality,
Hi,
if that means I should post my config, here goes:
zaptel:
span=1,1,3,cas,hdb3,crc4
cas=1-15:1101
cas=17-31:1101
unicall.conf:
protocolvariant=id,10,10
protocolend=cpe
group=1
channel = 1-15
channel = 17-31
wanpipe1.conf
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= CRC4
Hi!
Unfortunately did this stop Asterisk to register ny phones and trunk.
Did I put tit in the wrong place?
//Mattias
Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
//Mattias
At 03:53 2007-01-05, you wrote:
exten = _9070X./209,1,NoOP,SORRY CHARLIE
exten =
Biggest feature: You need a patent license to use the codec. The intel
software does not include a patent license.
Matthew Rubenstein wrote:
Thank you, that is excellent advice.
I understand that Intel has a free g729 codec, and that there might be
others. Free g729 codecs cheat
What about the free open source G729
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Matthew Rubenstein wrote:
I connect to a PSTN carrier over SIP which requires me to connect with
a g729 codec. I'm using them for just robocalling: Asterisk server
Hi there,
I want to add 4000 Callerids and Callernames to my asterisk-db.
(/var/lib/asterisk/astdb)
I do not want an external database or an sql-database because I do
not want asterisk to depend on external processes.
However, when I do 4000 database put number name via a shellscript
and
Al Bochter wrote:
What about the free open source G729
To use a g729 codec you must pay a license fee to the patent holder. It
is immaterial as to whether the implementation is open/closed source.
___
--Bandwidth and Colocation provided by
You know that if you rename an open Unix file, it will stay open - i.e. if
you rename the logfile full to full.1, Asterisk will continue writing
to full.1 thinking it was full.
The logger rotate command forces all log files to be closed and reopened
with their canonical names, so your file
Hello List,
I am curious how the ordering of the extensions are determined for an
ARA dial-plan. For example, if I have these:
_9X.
_9011.
Which is selected first? Any number dialed starting with 9011 is
matched by either rule here and I don't remember seeing any ORDER BY
clauses when
I am sot sure, but you can use the following to make sure:
_9.
_9011.
--
--
Steven
http://www.glimasoutheast.org
Jesse Peterson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hello List,
I am curious how the ordering of the extensions are determined for an ARA
Hi All,
In the realtime voicemail table the column 'customer_id' is used, for
my purpose, to specify the customers accountcode. The column name
'accountcode' is used in the iax and sip tables. To keep this
consistent throughout the tables, is there any reason I should NOT
switch the column
How much would all of that data slow down asterisk?
Is astdb made to handle that much data?
--
--
Steven
http://www.glimasoutheast.org
Christoph Adomeit [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi there,
I want to add 4000 Callerids and Callernames to my asterisk-db.
As far as I know, the g729 patent requires buying a license to operate
any implementation of it, whether Digium's, Intel's, or any other.
Digium is set up to collect royalties (perhaps at a favorable rate) as
part of their license from the patent holder. I don't know about Intel
or any
Hello,
How many licenses to buy?? :
From what we understood from digium website, we must buy as many
licenses as the number of maximum simultaneous calls using G729 Codec we
wish to make.
For example, If we want to be able to make a maximum of 10 simultaneous
calls using G729 Codec, we
Leo Ann Boon wrote:
Are you doing a blind transfer or attended transfer? I'm assuming
you're using the phone's transfer button. You may need to press
transfer a second time to complete the transfer.
Attended. I have decided my problem is with transfer, not specifically
parking. I just
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
Consult the wiki!
--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
Hosted IP PBX Services for SOHO Small
Yes
Zoa
Michel wrote:
Hello,
How many licenses to buy?? :
From what we understood from digium website, we must buy as many
licenses as the number of maximum simultaneous calls using G729 Codec
we wish to make.
For example, If we want to be able to make a maximum of 10
simultaneous
On Mon, Jan 08, 2007 at 10:51:03AM -0500, Al Bochter wrote:
What about the free open source G729
There's no such thing ... g.729 (as per the ITU specification) is patent
encumbered. Anyone USING the codec has to pay a license to the patent
holders.
Digium have negotiated a bulk-buying
Scott Walde wrote:
I have even caused Asterisk to crash a couple of times. I'm beginning
to wonder if I'm hitting a bug in 1.4. I've noticed a few other
people have got the notify answer... message. I'm going to try
downgrading to 1.2.14 to see if it works there.
In the meantime, and
What about the free open source G729
There's no such thing ... g.729 (as per the ITU specification) is patent
encumbered. Anyone USING the codec has to pay a license to the patent
holders.
I believe (this may have changed) that ANY patented technology can be used
for free educationally. The
One thing that is really confusing me at this point: if I want to
leave
an automated answer machine message, and amd tells me it's a machine,
how do I know when to start leaving the message ? Some intros are
long
(thanks for calling, me and mine are not here right now, please leave
a
message
That's not correct. You need one G729 license for each transcoding instance. If
you have two SIP channels and both are G729, then no license is required. If
you have two SIP channels, and one is G729 and the other is ulaw, then a
license is required.
Doug.
-Original Message-
From:
Someone know why my asterisk gives me the following msgs?
Thank you
- Got SIP response 603 Declined (no dialog) back from
X.X.X.Xhttp://82.51.224.34/
-- Got SIP response 603 Declined (no dialog) back from
X.X.X.Yhttp://82.51.224.34/http://82.104.4.192/http://82.104.4.192/
-- Got SIP
Hello!
Sorry for the OT-thread, but i don't know where else too ask...
Has anyone done http-provisioning of a Linksys SPA942 with client side
ssl-authentication? Where do i get the CA from?
I'm aware of the Sipura mass deployment howto on voip-info.org, but it
doesn't cover the authentification
First point to tackle in any case involving patent, copyright or
trademark infringement is whether or not the infringing party would have
been qualified to buy any usage rights at all. In a case where you
license the Intel source(read the terms, it's not really that free),
you would be applying
Hi,
I am looking for tollfree number in italy. Anybody providing that? Charge per
minute? It will connect to my asterisk pbx box.
Thanks
CM
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
Buki wrote:
Sorry I forgot to change the subject line in my last posting!
I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1
for many months now and I am a big fan and I have been very
happy with it.
I'm glad it's working well for you, positive feedback is always
welcome.
I
Greetings,
I am using MixMonitor to record my outgoing calls. It seems that
MixMonitor will not write to a directory if it doesn't exist (ie - it
doesn't create a new directory if needed).
I have checked to ensure permissions are properly set, and if I manually
create the directory,
Jerry wrote:
What about the free open source G729
There's no such thing ... g.729 (as per the ITU specification) is patent
encumbered. Anyone USING the codec has to pay a license to the patent
holders.
I believe (this may have changed) that ANY patented technology can be used
for
Mike,
So tell me what this FREE open source G729 is
I am told that you can use these Codecs with your Asterisk !
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
You can do it Freely !!
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Mike
On Mon, Jan 08, 2007 at 02:53:39PM -0500, Al Bochter wrote:
So tell me what this FREE open source G729 is
I am told that you can use these Codecs with your Asterisk !
[1]http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
You can do it Freely !!
No, Ready Technology
You need a license when ever you transcode the audio
From any codec to G729. or G729 to any codec
you will need a license for each instance.
If you call into your system from a provider that uses G729 you don't
need a license
If you check your voicemail that is saved on your system in GSM
On 1/8/07, Al Bochter [EMAIL PROTECTED] wrote:
Mike,
So tell me what this FREE open source G729 is
I am told that you can use these Codecs with your Asterisk !
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
You can do it Freely !!
Al, I don't know if you're stupid, or you
Performance/Price wise which implementation of the codec is better?
Digium or the Ready Tech/Intel IPP code?
I'm looking at building a 4 PRI g.729 Asterisk box (Dell 2 x dual
core, Digium 4 T1 + echo canceller). Which codec would provide the
best audio quality?
--
Matthew S. Crocker
Al Bochter wrote:
Mike,
So tell me what this FREE open source G729 is
I am told that you can use these Codecs with your Asterisk !
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
You can do it Freely !!
Please read the entire page. From the link you sent:
Why NOT G.729?
Matthew Crocker wrote:
I'm looking at building a 4 PRI g.729 Asterisk box (Dell 2 x dual core,
Digium 4 T1 + echo canceller). Which codec would provide the best audio
quality?
G.729 is G.729 (assuming same suffixes like B, C, etc.). Audio quality
is exactly the same, or the implementations
Mike
I understand that.
but it states on there site and note the key words may need
What I want to know is if you buy 10 licenses from digum can use the
Open Souce code?
As long as you don't transcode than 10 at a time. Is that legal?
I see the note about the IPP license
From what I have
Tzafrir Cohen wrote:
On Mon, Dec 11, 2006 at 12:11:34AM +0100, Andreas von Heydwolff wrote:
I'm using 1.2.13~dfsg-2 from Debian unstable in a small SOHO
environment, it's doing its job.
However, the startup scripts seem to hose something and it's running but
not working with
Sorry again Al but you are way off on this one also.
sipro licenses digium who licenses end users for the digium product they
are buying. It's like saying if I buy an ATA with 2 g.729 licenses can I
throw it away and use the licenses with my open source codec? No way!
Al Bochter wrote:
Mike
Al Bochter wrote:
Mike
I understand that.
but it states on there site and note the key words may need
What I want to know is if you buy 10 licenses from digum can use the
Open Souce code?
That is not what you said or asked. You were asserting that a free as
in beer solution existed. If
All of which hassle and expense can be avoided by buying a license for
Digium's codec, which is tested to work well with Asterisk (and might
come with some support). And is pretty cheap per simul call.
I wonder whether that per call means per codec instance, which
could be
per call means per terminating channel where encoding/decoding is
required. Termination could be to translate to another codec (with
another peer) or to Asterisk itself to handle menus, voicemail,
conference calls.
In the conference call setup, each caller uses a license.
-Original
Hey guys. This is the setup that I have for a voicemail account.
1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=yes
It emails me the voicemail, but it does not delete it from the system
afterwards. I have also tried
1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=1
Thought I might just as well share these scripts, they may work with
other phones too:
*1)* Dialing from the KDE 3.5.5 address book works with a script that
gets triggered from the kaddressbook (Settings - Script Hooks -
On 15:17, Mon 08 Jan 07, Mark Greene wrote:
Hey guys. This is the setup that I have for a voicemail account.
1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=yes
try this:
1509 = 1509,Mark Greene,[EMAIL PROTECTED],,attach=yes|delete=yes
You forgot about the pager field.
It
Anyone know the command that tells * to load a sipfriend
from the realtime db rather than saying no such host? I've tried various
combinations of the rt commands:
rtcachefriends=yes;
;rtcache=yes
;rtAutoClear=yes
;rtautoreg=yes
;rtIgnoreRegExpire=yes
;rtupdate=yes
I have googled and I do not understand how the pager field is what is
causing the problem.
Could you explain?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
G.729 is G.729 (assuming same suffixes like B, C, etc.). Audio quality
is exactly the same, or the implementations aren't compatible.
Yes, but depending on the implementation the CPU resources between
two could be quite different. Audio quality could be adversely
affected by inadequate
On 15:51, Mon 08 Jan 07, Mark Greene wrote:
I have googled and I do not understand how the pager field is what is
causing the problem.
Could you explain?
If you dont provide it the parser will think the pager
address is 'delete=yes|attach=yes'
--
Michiel van Baak
[EMAIL PROTECTED]
Mark Greene wrote:
I have googled and I do not understand how the pager field is what is
causing the problem.
Could you explain?
Think of it as a CSV file. The ,, entry for pager is just a placeholder
saying that for pager there is nothing. Omitting means that the next
field will be
Mike
What I was looking to do is use the easier to install one and the better
one.
I was asked by a customer about using G729 and I told the customer that
they would have to pay for the G729
licenses. The customer pointed out the open source G729 code and I was
not sure if I could use that.
Matthew
I agree. I only know what I have told by others so I do need this input
I have been told that Digum G729 is a big pain the the butt to get
working with Asterisk
and it is very hard on the CPU
Keep in mind I have never used any Ver. of G 729
So tell me what you think.
Best regards,
post here your extensions.conf
On 1/8/07, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi!
Unfortunately did this stop Asterisk to register ny phones and trunk.
Did I put tit in the wrong place?
//Mattias
Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
I did some tests a long time ago and the speed was roughly the same. ( I
think digium's was slightly faster).
I think the IPP version also doesn't work on AMD out of the box.
It's just 10$ a channel, that's not even worth the hassle of trying
something else.
Joachim
Al Bochter wrote:
Did you find any operations trouble installing/using the Digium codec
with Asterisk? I'd be surprised if Digium's were hard to use with
Asterisk, considering they wrote and support both. Also can their codec
be used to pre-encode data to files from a Linux command/line? Or just
the
[EMAIL PROTECTED] wrote:
In iax.conf there is option jitterbuffer
how about sip protocol ? Are jitterbuffer can configure in sip.conf ?
In 1.2 and 1.0 there is no jitter buffer for SIP. I think 1.4 might
have a SIP jitter buffer, but I'm not sure. Check sip.conf.sample in 1.4.
Lex Lethol wrote:
As far as I know when I setup a 3-way on something like a cisco will
disconnect everyone when the middle (person who setup the conference)
hangs up.
The problem I describe happens on ATAs and the like that uses flash to
put on hold while setting up the second call.
I am not
On 1/8/07, Al Bochter [EMAIL PROTECTED] wrote:
I agree. I only know what I have told by others so I do need this input
I have been told that Digum G729 is a big pain the the butt to get
working with Asterisk
and it is very hard on the CPU
Keep in mind I have never used any Ver. of G 729
I
The DISA application (show application disa at the CLI) will let you do
this.
Mojo
On Saturday, January 06, 2007 10:19 pm, Yuan LIU wrote:
After the user navigated some voice menus, how do I give him another (fake)
dial tone?
Yuan Liu
___
Hi All,
I am starting to get complains from users that the call volume is very
low and people are having problems haring what is said. This is for
internal calls (between extensions) and over ZAP. The problem seems to
be with the caller and callee, no matter if it is an incoming or
outgoing
Klaverstyn, David C wrote:
Hi All,
I am starting to get complains from users that the call volume is very
low and people are having problems haring what is said. This is for
internal calls (between extensions) and over ZAP. The problem seems to
be with the caller and callee, no matter
Thanks Eric, That's what I figured but I wanted to make sure that it was
not possible to increase VoIP volume levels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, 9 January 2007 9:35 AM
To: Asterisk Users Mailing
OK that makes a lot more sense. Thanks for the explanation.
- Mark
___
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To UNSUBSCRIBE or update options visit:
It sounds fairly easy to me.
If I had a 1.4 system built I would write something in perl to do that
and put it under gpl. You could also do it in php or anything else that
can be run from command line and has an asterisk manager interface
available.
I don't even need the g729 codec to get it
All SIP devices I have used allow you to increase the volume levels.
But it is done in the DEVICE, not in Asterisk.
Klaverstyn, David C wrote:
Thanks Eric, That's what I figured but I wanted to make sure that it was
not possible to increase VoIP volume levels.
Looks like indeed I was underclocked, new results:
[EMAIL PROTECTED]:~# cat /proc/cpuinfo
processor : 0
vendor_id : CentaurHauls
cpu family : 6
model : 10
model name : VIA Esther processor 1200MHz
stepping: 9
cpu MHz : 1197.301
cache size : 128
On Mon, Jan 08, 2007 at 04:59:58PM +0100, Christoph Adomeit wrote:
Hi there,
I want to add 4000 Callerids and Callernames to my asterisk-db.
(/var/lib/asterisk/astdb)
I do not want an external database or an sql-database because I do
not want asterisk to depend on external processes.
The Intel IPP-based G.729 codec does work with AMD processors out of the box,
both with the 32 bit and 64 bit versions.
On Mon January 8 2007 19:31, Zoa wrote:
I did some tests a long time ago and the speed was roughly the same. ( I
think digium's was slightly faster).
I think the IPP
Listen to the first 30 (or so) seconds of
http://pod-serve.com/audiofile/filename/4353/interviews-podcast-allison-smith.mp3
There's a queue-related sound file recorded by Allison: Thank you for
holding... You are the umm, you are SO FAR DOWN the list we won't get your
call today, probably
When I first noticed that this thread has over 20 messages i was sure
it is interesting. When I read it I realized that I havn't noticed
that Al Bochter has posted to it.
Plain old stuff, just someone making sure to put a new twist on it.
On 1/8/07, Juan Jose Comellas [EMAIL PROTECTED] wrote:
Good luck dealing with Linksys on that
http://voxilla.com/tools/device-configuration-wizard/certificate-authority-service-for-linksys-analog-voip-adaptors-808.html
On 1/8/07, Benko [EMAIL PROTECTED] wrote:
Hello!
Sorry for the OT-thread, but i don't know where else too ask...
Has anyone
On 1/8/07, lenz [EMAIL PROTECTED] wrote:
You know that if you rename an open Unix file, it will stay open - i.e. if
you rename the logfile full to full.1, Asterisk will continue writing
to full.1 thinking it was full.
The logger rotate command forces all log files to be closed and reopened
with
I do not think that there are some company that offer a toll free number
(Numero verde in italian)
But contact on of these three providers
http://www.eutelia.it/tlc/
http://www.unidata.it/
http://messagenet.it/
If they have one of these should be able to give to you
See you
On 1/8/07, CM
Thankyou David,
It works for Linksys,but not for snom 360.
Do I need to change someting using web UI ?
--- Klaverstyn, David C
[EMAIL PROTECTED] wrote:
This is my code (that I copied form somewhere) for
paging a group of
phones. By dialling 99 it will page phones 2101,
2102 and 2105.
Jerry wrote:
I believe (this may have changed) that ANY patented technology can be
used
for free educationally. The idea is that people can study and play with
the technology for no charge. I'm not sure if this means that a
University can use this in their phone system without paying the patent
I knew I was doing the right thing, here is the proof, enjoy when you
read it, and have a good laugh.
-- Forwarded message --
From: Al Bochter [EMAIL PROTECTED]
Date: Jan 8, 2007 8:22 PM
Subject: Re: [asterisk-users] Some queries on g729 license.
To: [EMAIL PROTECTED]
(C)UNT
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