Hi all,
I have spent some time searching, but I haven't found a way to prevent *
from concatenating two frames into one IAX packet.
I have a situation where I make an IAX GSM call to *, which transcodes
to an iLBC SIP call. Every second voice packet the IAX client receives
contains 2x 20ms
Chris Bagnall schrieb:
As someone else pointed out, the Siemens C450 IP (and higher models) work
great!
I should point out that for the relatively small price difference, it's well
worth getting the S450 rather than the C460.
We have a couple of S450 in production.
They basically work and
Hi all,
i am wondering if i can make two leg cdr in mysql cdr table.
1st Leg : Registrar the ATA which registered to the asterisk and it normally
logging in cdr table.
2nd Leg : The CDR of carrier for the example if i send call like
exten = _x.,1,Dial(SIP/[EMAIL PROTECTED]TIP)
I this cause
On 01:22, Thu 07 Feb 08, Abdul wrote:
Hi all,
i am wondering if i can make two leg cdr in mysql cdr table.
1st Leg : Registrar the ATA which registered to the asterisk and it normally
logging in cdr table.
2nd Leg : The CDR of carrier for the example if i send call like
exten =
On Wed, 6 Feb 2008 20:12:21 +0100, randulo [EMAIL PROTECTED]
wrote:
the phone referred to that Jared mentioned is the Allnet
7960. I have an ongoing review of it here (meaning I never finished it
properly).
Thanks for the tip.
___
-- Bandwidth and
On Wed, 6 Feb 2008 20:12:21 +0100, randulo [EMAIL PROTECTED]
wrote:
http://food4wine.ning.com/
BTW, we also want to receive call notifications on our cell phones. In
addition to using SMS, we found a cheaper alternative which is to use
iMode cellphones and subscribe to Bouygues Telecom's
On 7 Feb 2008, at 09:17, David Hogan wrote:
Hi all,
I have spent some time searching, but I haven’t found a way to
prevent * from concatenating two frames into one IAX packet.
I have a situation where I make an IAX GSM call to *, which
transcodes to an iLBC SIP call. Every second
On 7 Feb 2008, at 10:29, Vincent wrote:
On Wed, 6 Feb 2008 20:12:21 +0100, randulo [EMAIL PROTECTED]
wrote:
http://food4wine.ning.com/
BTW, we also want to receive call notifications on our cell phones. In
addition to using SMS, we found a cheaper alternative which is to use
iMode
Well alternatively you can look up straight forward gsm voip gateway.
Like CT-375
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Thursday, February 07, 2008 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 7 Feb 2008, at 00:36, Tilghman Lesher wrote:
On Tuesday 05 February 2008 09:22:29 Cavalera Claudio Luigi wrote:
Hello,
I'm doing some research concerning iax encryption, I haven't find any
clients (softphones or hardphones) which implement so I have not
tested
it yet.
There was also
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ash Rah
Sent: Wednesday, February 06, 2008 4:53 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TE412P and Delll PowerEdge 2900
Hello,
Looking for comments if Digium TE412P
What I am asking for is something to take an incoming SIP INVITE, change the
codecs listed in the SDP, forward the (new) INVITE to a media gateway,
perform the reverse codec handling for the 200 OK and perform RTP
transcoding on the resulting 2 legs of the call.
-How can asterisk do that !
-do
- No shared adress book (especially it should be shared between phone on
different base stations). I can access an online adress book, but only
the built in, and you cannot set up your own online book.
You can send address books to the phone in standard vcard format (though for
some reason it
G'day. I am wanting to find out how my SIP service is performing with
Asterisk, especially jitter and dropped packets.
I can get an overview of that using the 'rtcp stats' function at the
console, but is there any way to get those logged to a file or some
other permanent record?
Nothing in
On 2/7/08, Daniel Pittman [EMAIL PROTECTED] wrote:
G'day. I am wanting to find out how my SIP service is performing with
Asterisk, especially jitter and dropped packets.
I can get an overview of that using the 'rtcp stats' function at the
console, but is there any way to get those logged to
On Jan 22, 2008 12:22 PM, Steve Davies [EMAIL PROTECTED] wrote:
Based on some rapid checks, 7.1.30 firmware behaves in exactly the same way.
Cheers,
Steve
As a follow up, I just spoke with our UK snom distributor, EFL, and
they are discussing this with snom already. It seems that there has
On Thursday 07 February 2008 08:05:40 Yves Räber wrote:
Hello,
I'm having troubles while using the Goto function in a realtime
extension. Here is the error message :
-- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1)
-- Goto (script_13_0,s,1)
[Feb 7 13:24:21] WARNING[28666]:
I am facing the same problem.
I have noticed that (sometimes) under heavy load (70+ calls at 1GHz P4
with 1GB ram) some AGIs seem to run continously, and I have to kill
them manually.
I use 'core show channels verbose', and the duration is like 1 hr or
more, while my telco does not allow calls
Hi,
I tired to use the following configuration but still in cdr table i can see
only one record.
sip.conf
[444]
type=friend
username=444
secret=444
host=dynamic
nat=yes
context=vpstoteles
disallow=all
allow=all
Extentions.conf
[vpstoteles]
exten =
We're deploying an asterisk-based phone system at all of our branch
offices in an effort to eliminate long-distance costs incurred from the
constant branch to branch calls. We're using the Snom 300's at all
offices for the desk phones and X100P cards to interface to 2 analog
lines. I'm
Asterisk does not support that yet.
Zoa
rachid wrote:
Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I
Hi,
we are using the app_nvfaxdetect from Newman Telecom with Asterisk 1.4
and tried to build the trunk/next release 1.6 with this application, but
it failed (We are using fax stuff with iaxmodem/Hylafax).
I remember that we had the same issue switching from 1.2 to 1.4 and
someone made the
Atis Lezdins wrote:
On 2/7/08, Olivier [EMAIL PROTECTED] wrote:
Hi,
Is it possible and safe to split or balance outgoing calls to 2 different
sip-to-tdm voice gateways ?
I need 5 E1 ports and the boxes have 4 ports each.
Setup would be :
PSTN --1xE1-- Gateway1 ---2xE1 PBX
They could be reporting glare back from the phone because the mic or
ear volume is miss matched, normalize the volume on the phones.
On Feb 7, 2008 12:40 PM, Brent Davidson [EMAIL PROTECTED] wrote:
We're deploying an asterisk-based phone system at all of our branch
offices in an effort to
rachid wrote:
Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
It's
Kevin P. Fleming wrote:
zoa wrote:
Asterisk does not support that yet.
Yes it does, and it puts G.722 into the SDP the way that RFC3551
specifies. To the original poster: please read RFC3551 and you will
understand why G.722 appears in the SDP with an 8000 sample rate instead
of 16000.
Hi,
Do you really think Presence should be used to forward call to voicemail ?
My feeling is forwarding incoming calls to voicemail should remain a
different task as you could wish to remain unavailable for chat and still
reachable by phone.
As I can't see a way to define Presence status such as
zoa wrote:
Asterisk does not support that yet.
Yes it does, and it puts G.722 into the SDP the way that RFC3551
specifies. To the original poster: please read RFC3551 and you will
understand why G.722 appears in the SDP with an 8000 sample rate instead
of 16000.
--
Kevin P. Fleming
Director of
Which phones does it work with ?
I know it works with the S450, we don't supply the cheaper C series.
Can you allocate each phone its own address book or does it need to be shared
among all of them ?
Each phone's address book is independent. We just wrote a set of scripts to
allow clients
Mojo with Horan Company, LLC wrote:
So just tell your client to not /ask/ for 8kHz audio.
As Kevin just pointed out, apparently you do NOT have to tell your
client to ask for 8kHz audio. May I ask what client you are using?
Moj
___
-- Bandwidth and
From the RFC:
Even though the actual sampling rate for G.722 audio is 16,000 Hz, the
RTP clock rate for the G722 payload format is 8,000 Hz because that
value was erroneously assigned in RFC 1890
http://www.faqs.org/rfcs/rfc1890.html and must remain unchanged for
backward compatibility. The
On Thu, 7 Feb 2008 11:03:22 +, Tim Panton [EMAIL PROTECTED]
wrote:
Is that some form of push notification?
Yup, it comes with the same push mail feature found in BlackBerry.
Much cheaper than either sending SMS's or taking a 3G subscription.
Can't wait for Wimax or cellphones over TV
I think I have my echo problem solved, now i need to tackle the MWI. I
can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a
section from my sip.conf for my test phone:
[general]
context=internal
allowguest=no
allowoverlap=no
allowtransfer=yes
notifyhold=yes
bindport=5060
The Asterisk development team has released Asterisk version 1.4.18.
In response to a community request, in preparation for this release, the
development community held a release candidate period before making the official
release. Multiple people tested it out and reported issues. The release
Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
Below wireshak trace:
On 2/7/08, Olivier [EMAIL PROTECTED] wrote:
Hi,
Is it possible and safe to split or balance outgoing calls to 2 different
sip-to-tdm voice gateways ?
I need 5 E1 ports and the boxes have 4 ports each.
Setup would be :
PSTN --1xE1-- Gateway1 ---2xE1 PBX TDM phones
Hello , i'm trying to found the issue that i have with the Authenticate
I'm using asterisk and cisco GW when i originate the call from ASterisk to
dial one number and it send to GW Cisco everything is ok with the Authenticate
, I listen one ring back and the dead air then 15 seconds i got
On 2/7/08, Abdul [EMAIL PROTECTED] wrote:
Hi,
I tired to use the following configuration but still in cdr table i can see
only one record.
sip.conf
[444]
type=friend
username=444
secret=444
host=dynamic
nat=yes
context=vpstoteles
disallow=all
allow=all
Extentions.conf
Hi,
Is it possible and safe to split or balance outgoing calls to 2 different
sip-to-tdm voice gateways ?
I need 5 E1 ports and the boxes have 4 ports each.
Setup would be :
PSTN --1xE1-- Gateway1 ---2xE1 PBX TDM phones
|
LAN
Hello,
I'm having troubles while using the Goto function in a realtime
extension. Here is the error message :
-- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1)
-- Goto (script_13_0,s,1)
[Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel
'SIP/siemens1-081f56b0' sent into
Hi,
I tired to use the following configuration but still in cdr table i can see
only one record.
sip.conf
[444]
type=friend
username=444
secret=444
host=dynamic
nat=yes
context=vpstoteles
disallow=all
allow=all
Extentions.conf
[vpstoteles]
exten =
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Khaled Chehab
Sent: 07 February 2008 12:33
What I am asking for is something to take an incoming SIP
INVITE, change the codecs listed in the SDP, forward the
(new) INVITE to a media gateway, perform the reverse codec
Olivier wrote:
At the opposite, I think it could be useful for an Asterisk server to
act as XMPP User Activity provider (ie update XEP-0108 field with
on-the-phone value).
Do you agree ?
Is there any XMPP client supporting User Activity ?
Is Asterisk capable of getting or sending such User
On Thu, Feb 07, 2008 at 07:53:12PM +, Ben Willcox wrote:
Olivier wrote:
At the opposite, I think it could be useful for an Asterisk server to
act as XMPP User Activity provider (ie update XEP-0108 field with
on-the-phone value).
Do you agree ?
Is there any XMPP client supporting
Hi Olivier,
At the opposite, I think it could be useful for an Asterisk server to act as
XMPP User Activity provider (ie update XEP-0108 field with on-the-phone
value).
Do you agree ?
This is indeed a direction we should consider in order to relay call
and device state information to XMPP
2008/2/7, Chris Bagnall [EMAIL PROTECTED]:
- No shared adress book (especially it should be shared between phone on
different base stations). I can access an online adress book, but only
the built in, and you cannot set up your own online book.
You can send address books to the phone
I would have been happy ... but it's not that. This query gives me the
right row (I double checked).
On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote:
On Thursday 07 February 2008 08:05:40 Yves Räber wrote:
Hello,
I'm having troubles while using the Goto function in a realtime
Philipp Kempgen wrote:
The cluster capability is very interesting.
Exactly. Although I must admit wo don't really have much
documentation on how to install a Gemeinschaft cluster.
Is there anything documented in English?
Philipp's mail doesn't say what the product does, and the website
Just something I noticed: your third line from extensions.conf begins with
s, while the other two begin with _X.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us at
www.generationd.com http://www.generationd.com/
Nevermind... I found the appropriate mojo. The key was putting
[EMAIL PROTECTED],password and removing the subscribemwi=yes. (I
think that's all it required. Among the other 1500 things I've already
tried, there may have been some residual.)
Thanks,
Brent
.
Brent Davidson wrote:
I think I
On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote:
I would have been happy ... but it's not that. This query gives me the
right row (I double checked).
On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote:
On Thursday 07 February 2008 08:05:40 Yves Räber wrote:
Hello,
I'm having
Hi there, I am looking to buy an Asterisk Appliance or Box for my organization
and was hoping to ask for recommendations. My ideal box is a small device in
size like Digium's AA50 Asterisk Appliance (
http://www.digium.com/en/products/appliance/ ) but will still have these
technical features :
Brent Davidson wrote:
I think I have my echo problem solved, now i need to tackle the MWI. I
[15]
mailbox=15
Should be [EMAIL PROTECTED]
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor
Brian Capouch wrote:
Philipp Kempgen wrote:
The cluster capability is very interesting.
Exactly. Although I must admit wo don't really have much
documentation on how to install a Gemeinschaft cluster.
Is there anything documented in English?
Not really. :(
But most of the comments in
I
would
have
been
happy
...
but
it's
not
that.
This
query
gives
me
the
right
row
(I
double
checked).
Make sure you don't have any labels on the prioritys. When loading extensions
from realtime labels aren't supported.
Replace:
exten = _X.,1(mylabel),...
with
exten =
Alternatively you could fix the client :-)
Heh :) Although it's a situation that won't happen in (our) production,
for the sake of completeness I'll probably upgrade the client.
Cheers,
Dave
___
-- Bandwidth and Colocation Provided by
On Feb 7, 2008, at 2:07 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Thu, Feb 07, 2008 at 07:53:12PM +, Ben Willcox wrote:
Olivier wrote:
At the opposite, I think it could be useful for an Asterisk server
to
act as XMPP User Activity provider (ie update XEP-0108 field with
For about 10 months I have been running a succession of Asterisk SVN
trunk versions on an Athlon 64 X2 4400+ based machine with OpenSuSE 10.2
at my home. I have a variety of SIP phones (mostly Polycom) internally;
my external connections are two POTS lines on a TDM400P (with HPEC) and
an IAX2
Astlinux on your own built box?
PaulH
On Thu, 2008-02-07 at 14:11 -0800, John Constalgie wrote:
Hi there,
I am looking to buy an Asterisk Appliance or Box for my organization
and was hoping to ask for recommendations.
My ideal box is a small device in size like Digium's AA50 Asterisk
jaap,
found this some time ago... might do the trick...
daveC
http://www.venturevoip.com/vm.pdf
Jaap Winius wrote:
Hi list,
After wrestling with the voicemail system for a while (Asterisk
1.4.14, Debian etch), I got it to work, but I still have lots of
questions, like:
* Why
Dear all
I am going to setup Asterisk Call center solution and i have
setup my queue and agent i have 2 SNOM ip phone but when i call to queue my
agent phone is rining without musicnhold or when both phone is busy then i call
to queue its directy hangup without musicnhole means
* Version: Asterisk 1.4.14
* Commas instead of pipes = already tried, this is not working at all
* Realtime switch for script_13_0 = No, should I ? This would be really
a shame, I want to use realtime BECAUSE I don't want to play with my
extensions.conf file. (I'm building a web interface that
I'm not using labels at all (but I've also tried with :))
On Thu, 2008-02-07 at 16:39 -0800, Grey Man wrote:
Make sure you don't have any labels on the prioritys. When loading
extensions from realtime labels aren't supported.
Replace:
exten = _X.,1(mylabel),...
with
exten =
This Friday as usual at 12 Noon EST, 9 AM PST, 17:00 UTC
Staying in the IVR domain for one more week, our guest today is
Allison Smith who literally needs no introduction.
I have a couple of secret tapes of Allison that she knows nothing
about from the earliest days of her career.
Instructions
Hi,
I am trying to communicate H323 and SIP users. I have configured h323.conf,
sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to
call successfully to h323 users using SJphone. And same for SIP users also.
But when I disabled gatekeeper and trying to call using
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