Hi,
I searched the archives about this, but didn't find any references.
When I make an outbound SIP call, the call completes and everything is
fine, but in the Asterisk console, I keep getting a huge stream of
warning messages:
WARNING[1200876848]: File dsp.c, Line 1107 (ast_dsp_process):
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Hi,
No matter what I configure my spans at (on a E400P) ztcfg -v always shows:
SPAN x: D4/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Currently I've configured my spans as ccs,hdb3,crc4, so shouldn't D4/AMI be
showing ccs/hdb3 instead?
- --
I have been on vacation so didn't jump in earlier. Some of what I say
here has been gone over earlier in this thread but I will repeat the
results as a summery.
AMI is not lossy, but it is almost always used in conjunction with a
ones density technique called bit7. Bit7 will change bit 7 to a
Comment out dmtfmode=inband or change it to something else.
With low-bandwidth voice codecs we don't have a good chance to decode
DTMFs, etc.
Martin
On Mon, 2 Jun 2003, Paul Cheng wrote:
Hi,
I searched the archives about this, but didn't find any references.
When I make an outbound SIP
Even after you reload the modules for the board ?
What about ztcfg -vv ?
Martin
On Mon, 2 Jun 2003, Tais M. Hansen wrote:
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Hi,
No matter what I configure my spans at (on a E400P) ztcfg -v always shows:
SPAN x: D4/ AMI Build-out: 0 db
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On Monday 02 June 2003 16:25, Martin Pycko wrote:
No matter what I configure my spans at (on a E400P) ztcfg -v always
shows:
SPAN x: D4/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Currently I've configured my spans as ccs,hdb3,crc4, so
hi All,
Since the quality drops with the increased
usageof IAX channels, between 2 Asterisk servers,
I want to limit the number of simultaneously used
IAX channels.
ie basically to limit the calls between the 2 Asterisk servers,
can anybody pls tell me a method/hint to achieve this, i am
hi surajee,
some time ago martin pycko posted an example to the list
how to make use of global variables and gotoif to limit the
number of channels.
the ML archive is your friend :)
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
Line 20 of channels/iax.h it has:
#define AST_IAX_MAX_CALLS 32768
This defines the total number of active
calls. You should be able to change it to whatever you like.
This would be a nice option to include in iax.conf.
-wade
-Original Message-
From:
[EMAIL
On Mon, 2003-06-02 at 10:16, Wade Weppler wrote:
Line 20 of channels/iax.h it has:
#define AST_IAX_MAX_CALLS 32768
This defines the total number of active calls. You should be able to
change it to whatever you like.
This would be a nice option to include in iax.conf.
The T400P/E400P works in both 3.3V and 5V configurations.
Mark
On Sun, 1 Jun 2003, Benjamin Miller wrote:
I second the question and request.
There are more and more server class machines that won't take the old PCI cards at
all.
-Original Message-
From: Gene Kochanowsky
Greetings...
I'm working on getting an AudioCodes Mediant 2000 big box o' PRI's going
with Asterisk, and am running into a problem with DTMF handling.
The box is sending DTMF packets to Asterisk as INFO packets, and they are
apparently being seen by Asterisk. However, the DTMF knowledge
Has anyone done anything with Asterisk using the
G.711 codec?
Also, is there a uncompressed option so that you
could assign a single port to be unconpressed audio?
Thanks!Stu
I did search in the list and i googled the web, but i couldn't find anything
related to my problem, if u hav that post can u pls send it to me,
or at least gv me some key words to search for..
Surajee
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL
I use g711 ulaw, works fine, g711 is uncompressed (64k) as far as I'm aware.
Regards
MIKE
Has anyone done anything with Asterisk using the G.711 codec?
Also, is there a uncompressed option so that you could assign a single
port to be unconpressed audio?
Thanks!
Stu
The T400P and T100P cards do work in PCI-X slots. Or at least on the speed selectable PCI-X slots
in Dell 2650 servers.
Bill
Gene Kochanowsky wrote:
Does anyone know if there are any plans for Zapata or anyone else for that matter to come out with a 3.3v PCI version or PCI-X version of those
Hello:
I would like to know what kind of hardware is needed to have a good
performance with asterisk for 12 concurrent VoIP calls ?
Also if anyone had used the Adtran Total Access 750 can you give me
some opinions about it ?
thanks,
Ivan
___
On Mon, 2003-06-02 at 11:20, Iván Aponte wrote:
Hello:
I would like to know what kind of hardware is needed to have a good
performance with asterisk for 12 concurrent VoIP calls ?
This question should really be looked at in the archive. This comes up
just about as often as new users
Also if anyone had used the Adtran Total Access 750 can you give me
some opinions about it ?
-quite stable, factory default setting work out of the box with zaptel T1
h/w
pass's callerid etc
-any firmware version prior to L34 would not properly support far end
disconnect supervision for me
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On Monday 02 June 2003 16:25, Martin Pycko wrote:
Even after you reload the modules for the board ?
What about ztcfg -vv ?
ztcfg didn't handle anything but T1 types in printconfig. Here's a quick patch
to fix this behaviour.
- --
Regards,
Tais
Hi all,
Here another guy working with ztdummy and having
problems with music.
MP3Player does not work for me both from a
telephone entering through a passive ISDN-adapter as well as a SIP-client in the
LAN. Ztdummy works with conferences.
Here are my messages.
I have seen several
I've been trying to use net2phone's sip service at sip.net2phone.com
with * but keep getting
SIP/2.0 401 Unauthorized. Do you know if this should be possible?
So far:
I can use an ata186 to connected directly to n2p through
sip.net2phone.com without any special settings.
I can connect from * to
Hello,
With some trepidation I've come to inquire about platform requirements for *
after having spent a couple of hours searching and browsing the archives and
skimming the Handbook (very nice). I've found recommendations for 800-1000
Mhz and 128-256 MB RAM machines. My curiosity is not about
On Monday 02 June 2003 03:46 pm, Mike M wrote:
With some trepidation I've come to inquire about platform
requirements for * after having spent a couple of hours searching
and browsing the archives and skimming the Handbook (very nice).
I've found recommendations for 800-1000 Mhz and 128-256 MB
On Mon, 2 Jun 2003, Mark Thompson wrote:
I can use an ata186 to connected directly to n2p through
sip.net2phone.com without any special settings.
I can connect from * to iconnecthere, but, whatever I try from * to n2p
produces SIP/2.0 401 Unauthorized
(Can forward the full * sip log and
On Mon, 2 Jun 2003, Tilghman Lesher wrote:
First, they're going to have to be MMX. The 133 might be, but the 75
is definitely not. Normally you don't want to go much below a 200MHz
processor for a base-level system; you could certainly try something
slower, but not without MMX
I find the example to predictive dialing or progresive dialing.
I wish to do webpage used by extensions or operators..
where them says click ...and the asterisk dial to a customer and the
extension same time.
(If the call is answered).
Thanks.
___
Thanks Stephen...
I am using linux terminal server (without hard disk) ... and works fast because
I am using a server Xeon 2.7 Ghz.
Now I am trying ...run voice ip in my terminals.
Stephen Davies wrote:
On Mon, 2 Jun 2003, Tilghman Lesher wrote:
First, they're going to have
Stephen Davies wrote:
On Mon, 2 Jun 2003, Mark Thompson wrote:
I can use an ata186 to connected directly to n2p through
sip.net2phone.com without any special settings.
I can connect from * to iconnecthere, but, whatever I try from * to n2p
produces SIP/2.0 401 Unauthorized
(Can forward the full
Asterisk-people,
Some of you may have heard that we were working on a simple, cross-platform IAX client library called iaxclient.
We've pretty much been on vacation with the project for a while, but recently have made some progress, and now have the library working across platforms, and
On Monday 02 June 2003 18:47, Scott Lambert wrote:
On Mon, Jun 02, 2003 at 04:46:51PM -0400, Mike M wrote:
Hello,
With some trepidation I've come to inquire about platform requirements
for * after having spent a couple of hours searching and browsing
the archives and skimming the
PRI is shorthand for Primary Rate Interface ISDN
BRI is Basic Rate Interface ISDN
John Harragin wrote:
Thanks to everyone for all the info...
One more question. Until now, I had succesfully avoided isdn throughout my
computer career... other than having the notion that it was a fairly
troublesome
I can use a global variable to keep a count of number of channels used on a
particular channel,
and i can increase this count when i am calling the 'Dial' application, but
the problem is, how can i
reduce the counter(when a channel is free), i can not find a place in dail
plan to do 'counter--'
Does it also support PCI-X?
-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED]
Sent: Monday, June 02, 2003 11:24 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Zapata 3.3v PCI version...
The T400P/E400P works in both 3.3V and 5V configurations.
Mark
On Sun, 1 Jun
Hi all,
I have a problem on starting the ISDN PRI T1 line using on Wildcard
T400P.It keeps saying the D-Channel of the span is down after T200
timeup and sending SADME and receive DM from network side.
How can I start the D-Channel? I have checked with the network side and
matched with their
Thanks!
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, June 02, 2003 12:19 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zapata 3.3v PCI version...
The T400P and T100P cards do work in PCI-X slots. Or at least on the speed selectable
PCI-X
Then use AGI.
Jeremy McNamara
Surajee Ratnayake wrote:
I can use a global variable to keep a count of number of channels used on a
particular channel,
and i can increase this count when i am calling the 'Dial' application, but
the problem is, how can i
reduce the counter(when a channel is
On Mon, 2003-06-02 at 21:58, Surajee Ratnayake wrote:
I can use a global variable to keep a count of number of channels used on a
particular channel,
and i can increase this count when i am calling the 'Dial' application, but
the problem is, how can i
reduce the counter(when a channel is
Thank you very much!
i was exactly looking for that
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 9:16 AM
Subject: Re: [Asterisk-Users] Limit Concurrent IAX Channels
On Mon, 2003-06-02 at 21:58, Surajee Ratnayake
Hello !
I made some changes in the extension.conf to make *
ask for a key when a person enters a conference room.
But it is not asking for any key and takes the user
directly to the conference room. I am using sip
softphones as end points connected to sip for
conferencing. Any pointers for the
hi!
I wanna do some arithmatic operations (addition and substraction -integer
operation) inside extensions.conf. Is there a simple way to do this. If I do
yy = ${xx} + 1 // say xx is initialized to '0'
the resulting yy will show
0 + 1
Obiviously not the result I need. Any help
We're getting ready to ditch our hosted virtual PBX for an Asterisk
solution and I've got a couple questions that probably come from
experience; we're looking to host two PBXs on the same box if the PBXs
are identified by different DID numbers, and I was curious if:
1. Is it possible to have
denzel:
read the asterisk/README.variables
exten = s,1,SetVar(v2=0)
exten = s,2,Playback(beep)
exten = s,3,SetVar(v2=$[${v2} + 1])
exten = s,4,GotoIf($[${v2} 2]?t|1:*|1)
line 3, increments the value of the variable, we use it to loop a context
for a limited number of times, etc...
- wasim
On Mon, 2 Jun 2003, Jason Smith wrote:
We're getting ready to ditch our hosted virtual PBX for an Asterisk
YAY! more power to ya...
1. Is it possible to have duplicate extensions between the two PBXs?
Eg: 555-x100 and 555-x100 on same * server
don't see any reason why
Hi all,
Maybe something is wrong with my mpg123 (0.59r).
When I make a wav file of sample-hold.mp3 with mpg123 -w x.wav -r 8000
sample-hold.mp3 I see that the 28 seconds of the original clip has been
shortened to half of it (14 seconds). The sound recorder is playing recorded wav
files
The problem with using Voice Modems is that they fall into two categories:
1) Hardware Modems which only have half-duplex transmission of voice
2) Soft/Win/Lin modems which are proprietry and don't have asterisk
drivers
or 3 - full duplex real voice modems such as produced by Banksia in
On Tue, Jun 03, 2003 at 05:47:54PM +1000, Mathew Frank wrote:
Woody wrote:
The problem with using Voice Modems is that they fall into two categories:
1) Hardware Modems which only have half-duplex transmission of voice
2) Soft/Win/Lin modems which are proprietry and don't have asterisk
Hello,
Is there some way to explain to * that must send all non local (or locally
unknown) numbers to a default SIP proxy?
For example, i'm using * locally with a few SIP phones but i register my * to
an external Vocal or any other SIP proxy. Can i send all non local calls to
that external
The special wav49 ms hack doesn't produce correct WAV files on output.
they cannot be played back on windows media player nor winamp.
for example a typcial voicemail message with 9 seconds stops after 6
seconds playback with the error: invalid fileformat (Error=8004022F)
attached is such a
Hi all,
There was something wrong. With mpg123 (pre0.59s)
there is a new option make linux-pentium. This produces wav files with the
correct length and now I can hearthose in players. Next step: Try out in
*.
Regards Jan.
- Original Message -
From:
Jan Boon
To:
Hi all,
Unfortunately no luck:
-- Executing MP3Player("Modem[i4l]/ttyI0",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stack
NOTICE[131084]: File app_mp3.c, Line 80 (timed_read): Selected timed
out/errored out with 0
-- Timeout on Modem[i4l]/ttyI0
-- Executing
On 2 Jun 2003 at 14:35, Brancaleoni Matteo wrote:
Little summary what the status is:
1. isdn4linux for real calls not just IVR forget it
2. capi4linux only works with cards active AND passive which do support capi
at the moment those are eicon diva and cards from AVM
The only passive card at the
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Hi,
I'm attempting to hook up the E400P card to a Siemens ISDN module.
I have no knowledge of these Siemens products, so I'm acting on what I've been
told about it.
The Siemens side is configured to ISDN30: ECMA QSIG. The Siemens manual
states
-- Executing Dial("SIP/sipphone-b6e6", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-ab35 answered SIP/sipphone-b6e6 --
Attempting native bridge of SIP/sipphone-b6e6 and
SIP/216.52.153.207-ab35
is what shows up on the console window
...
thanks
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On Tuesday 03 June 2003 12:52, Tais M. Hansen wrote:
The Siemens side is configured to ISDN30: ECMA QSIG. The Siemens manual
states the card provides 30 ISDN B-channels which can be used for trunking
or networking.
Okay, I've done even more
Hi to all,
I've just received my Cisco 7905G ipphone. I want to connect it to asterisk
server but it looks that it has been preloaded with sccp protocol, so I
guess I need H.323 or SIP firmware image of some kind. I have a working tftp
server on my asterisk box alsoWhat do I need to do now to
Hallo Klaus-Peter,
i am currently working on a zaptel BRI driver which will support the
very nice (also nicely priced) hfc-pci based cards and the multiBRI
cards. the driver will support TE and NT mode.
Any timeplan?
I'm considering buying a diva server card with echo cancellation but I'd
more about the same problem ...
i've been playing around and got to this error
message which seems relevant ..
*CLI dial 1303 --
Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-1fb9 answered OSS/dsp Console call has been
sorry i'm sending so many emails, I always think of
something
exactly after i've pressed Send .. please be
patient with me :)
I also have OH323 installed, supposedly correctly,
and the same
gateway I want to connect to on SIP also supports
H323, however
i do not know what the dialcommand
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