David T Hollis wrote:
I'm not saying that it's chan_oh323's fault, but using the
openh32/pwlib CVS with chan_oh323 0.5.1 segfaulted but when I built
chan_oh323 against the latest release revs of openh323 and pwlib, it
worked fine. It's entirely possible that the openh323 CVS code is not
real s
On Fri, Mar 14, 2003 at 03:44:46PM -0500, John Vozza wrote:
> Thanks for the feedback but I'm still lost on this one (Forgive my
> ignorance please)
>
> I don't understand how "#define AST_FORMAT_ADPCM(1 << 5)" becomes
> a "format = 32" in the * console display.
2*2*2*2*2 = 32.
Okay
No, since the echo happens even when calling locally. Should I turn it
on for the T100P (analog phones attached to channel bank)?
Thank,
Mike
On Friday, March 14, 2003, at 12:18 PM, Richard Tomson II wrote:
Have you turned on the echo cancellation on the X100P?
Check the zaptel Makefile. I use
Since chan_capi now supports echo_cancellation on (some) eicon cards, i'm
considering buying another isdn-card.
On the moment I'm using an old Teles card with isdn4linux, but i get a
terrible echo when calling analog counterparts, and the delay is also quite
heafty.
1. If I get a (cheap) AVM-card
Jeremy McNamara wrote:
David T Hollis wrote:
I had the same problem, though asterisk wouldn't even start. It would
bomb out when loading the chan_oh323 module. The solution was to not
use the CVS distro just yet, there must be something not quite right
at this point. Go to the stable releases
Binary mathematics You can think of 1<<5 as 2 to the 5th
power.
1<<0 be 1
1<<1 be 2
1<<2 be 4
1<<3 be 8
1<<4 be 16
1<<5 be 32
Perhaps there might need to be a slightly more friendly message, but
remember its just a debug message.
Jeremy McNamara
John Vozza wrote:
Thanks for
On Fri, 2003-03-14 at 14:44, John Vozza wrote:
> Thanks for the feedback but I'm still lost on this one (Forgive my
> ignorance please)
>
> I don't understand how "#define AST_FORMAT_ADPCM(1 << 5)" becomes
> a "format = 32" in the * console display.
1 << 5
means to shift 1, 5 bits t
Thanks for the feedback but I'm still lost on this one (Forgive my
ignorance please)
I don't understand how "#define AST_FORMAT_ADPCM(1 << 5)" becomes
a "format = 32" in the * console display.
Regards
John
-
Net
On Fri, 2003-03-14 at 14:14, Zach Mesel wrote:
> Hi there.
>
> My wife and I run a small business out of our home. We have 2 phone
> lines, and don't currently need any additional extensions. What I and
> am in great need of is a voicemail system with the following
> functionality:
>
> - When w
Have you turned on the echo cancellation on the X100P?
Check the zaptel Makefile. I use the Mark2 with Aggressive_suppressor.
Rich.
On Fri, 2003-03-14 at 10:42, Mike Reiling wrote:
> Hello everyone,
> I finally got my Cisco 7960. Seems like a very nice phone. I have it
> configured for S
Hi there.
My wife and I run a small business out of our home. We have 2 phone
lines, and don't currently need any additional extensions. What I and
am in great need of is a voicemail system with the following
functionality:
- When we don't pick up one of our regular phones, the system
should
On Friday 14 March 2003 12:46, Michael Bielicki wrote:
> why don't u switch dns off on the cisco ?
Because i like to call hosts by name :)
Anyway, the DNS lookups are not the cause for the delay (tried disabling dns,
no more queries, but still a .5s delay in RTP)
lele
__
On Friday 14 March 2003 01:08 pm, Roderick Montgomery wrote:
> The list archives at http://www.marko.net/asterisk/archives/>
> seem to've stopped archiving message on Feb 13. Will the new
> messages to asterisk-dev and asterisk-users be in a separate
> archive, or the old message somehow merged int
>I have T working here.
Is your definition of T, that the caller can transfer ??
ie I P/U Zap 1 call Zap 2, & Now I press #701 & I can Park Zap2
If so is this a patch ??? pls post
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.
Sounds like a good idea to me. Some of the builtin help still refers to
using BYEXTENSION as well so that needs to be changed too
James
On Fri, 14 Mar 2003, Mark Spencer wrote:
> Perhaps we should have BYEXTENSION print a warning that says the option is
> deprecated, what do you think?
>
> Ma
David T Hollis wrote:
I had the same problem, though asterisk wouldn't even start. It would
bomb out when loading the chan_oh323 module. The solution was to not
use the CVS distro just yet, there must be something not quite right
at this point. Go to the stable releases and you will probably be
The list archives at http://www.marko.net/asterisk/archives/> seem
to've stopped archiving message on Feb 13. Will the new messages to
asterisk-dev and asterisk-users be in a separate archive, or the old message
somehow merged into a new archive?
I know it's not a top priority, but having a search
On Fri, 2003-03-14 at 12:11, Mark Spencer wrote:
> Perhaps we should have BYEXTENSION print a warning that says the option is
> deprecated, what do you think?
I wouldn't be opposed to it.
/me runs to see if I'm still using BYEXTENSION in my important configs.
--
Steven Critchfield <[EMAIL PRO
Hello everyone,
I finally got my Cisco 7960. Seems like a very nice phone. I have it
configured for SIP and working with asterisk. But, I am having some
echo problems. The person I call does not hear an echo, but I hear my
own voice echo back to me about .5 seconds later (maybe a little les
I have T working here.
--On Friday, March 14, 2003 9:40 AM -0600 Martin Pycko <[EMAIL PROTECTED]>
wrote:
Of courese:
exten => 9998,1,Dial,SIP/9998|30|tTm
Notice when you don't use the timeout you do have to use the options
separator "|" like this:
exten => 9998,1,Dial,SIP/9998||tTm
but I think t
Perhaps we should have BYEXTENSION print a warning that says the option is
deprecated, what do you think?
Mark
On 14 Mar 2003, Steven Critchfield wrote:
> On Fri, 2003-03-14 at 10:22, Don Pobanz wrote:
> > We have a group of lines (FXO/FXS) between our Rolm PBX and our
> > Asterisk server. From
The formats that asterisk uses are #define'd in
asterisk/include/asterisk/frame.h
RTP formats are #define'd in asterisk/rtp.c
regards
Martin
On Fri, 14 Mar 2003, John Vozza wrote:
> I've been trying to find a list of codec "format numbers" so I can more
> clearly understand the following messag
I know this may seem like a stupid question but how do
I compile gnophone, I am using RH8.0 and as I have experienced
and read in the mailing list it segfaults when the RPM is used..
So I have checked out the CVS source..here is a list of the files..
acconfig.h ChangeLog CVS gnophone.sh
Title: Message
Hi
All,
I am now a true
believer in Asterisk... I just made a call which went like
this:
Analog Phone
<-> ATA-186 <--SIP--> home asterisk <--IAX over DSL
1024/512--> office asterisk <-- digium E1 --> PSTN <--> gsm
cellphone
whilst the gsm user
who had no idea of all c
> This is what I have. In the MGCP box you have a few settings to deal with
> as well, of course. Currently MGCP is not functioning well. The "guys" are
> working on it. Right now, * will seg fault when you place station to
> station calls.
Already fixed in CVS as of this morninga ctually.
Ma
I've been trying to find a list of codec "format numbers" so I can more
clearly understand the following message;
Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx, requested format = 4,
actual format = 4
I've seen 4, 32, 512 and I think a few others. For example I think format
32 equal ADPCM but
That did it!
By using
${EXTEN}
instead of
BYEXTENSION
My call detail record now shows "ZAP/g1/xxx" where xxx is my extension
number.
Thanks.
Don Pobanz
On Friday, March 14, 2003 10:56 AM, Steven Critchfield
[SMTP:[EMAIL PROTECTED] wrote:
> On Fri, 2003-03-14 at 10:22, Don Pobanz wrote:
> > W
Liaan,
in your modules.conf you have to put:
noload => chan_h323.so
;noload => chan_oh323.so
if you would like to test the other on, just comment it :)
Lubo
Liaan van der Merwe wrote:
Ok... will try this
But how do is tell asterisk to use this channel and not default..
???
-Original Message
On Fri, 2003-03-14 at 10:22, Don Pobanz wrote:
> We have a group of lines (FXO/FXS) between our Rolm PBX and our
> Asterisk server. From the asterisk server any extension can be dialed
> regardless of system. Asterisk will then either ring the appropriate *
> extension or will dial a line into o
We have a group of lines (FXO/FXS) between our Rolm PBX and our
Asterisk server. From the asterisk server any extension can be dialed
regardless of system. Asterisk will then either ring the appropriate *
extension or will dial a line into our Rolm PBX and dial the
appropriate Rolm extension.
This is what I have. In the MGCP box you have a few settings to deal with
as well, of course. Currently MGCP is not functioning well. The "guys" are
working on it. Right now, * will seg fault when you place station to
station calls.
[clarent]
host = 192.168.xxx.xxx
context = default
line => aa
Of courese:
exten => 9998,1,Dial,SIP/9998|30|tTm
Notice when you don't use the timeout you do have to use the options
separator "|" like this:
exten => 9998,1,Dial,SIP/9998||tTm
but I think that T is not yet implemented
regards
Martin
On Fri, 14 Mar 2003, WipeOut . wrote:
> Thanks the 'show app
On Thu, 2003-03-13 at 15:45, Christoph Schütz wrote:
> Hello,
>
> I want to start a xmessage box if a call comes in.
> Is there a possibility to start a programm as the first extension?
> Like:
> exten => s,1,xmesage ...
> or something like that?
You may want to find a better way of getting that
Thanks the 'show application dial' was useful..
Can multiple options be specified?
eg. exten => 9998,1,Dial,SIP/9998|30|t|T
- Original Message -
From: Pertti Pikkarainen <[EMAIL PROTECTED]>
Date: Fri, 14 Mar 2003 15:15:14 +0200
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to
Hey guys, the occam people are coming in Monday with boxes for us to
test, they are controlled via MGCP, I've never used MGCP with asterisk
(or ever) and haven't been able to find much info about them on the
list. My guestimate config is as follows, let me know if I'm on the
right track.
mgcp.conf
anyone has the 't' option working with current cvs?
I can't make it work. but remeber it was working
(don't know what cvs)
Matteo
Il ven, 2003-03-14 alle 14:15, Pertti Pikkarainen ha scritto:
> I have it like this
>
> exten => 9998,1,Dial,SIP/9998|30|t
>
> 30 is a timeout value
> Check 'show a
I have it like this
exten => 9998,1,Dial,SIP/9998|30|t
30 is a timeout value
Check 'show application dial'
WipeOut ™ wrote:
What is the correct syntax to use the 't' option??
This is the current line in my extensions.conf
exten => 9998,1,Dial,SIP/9998
So would I change it to
exten => 9998,1,Di
What is the correct syntax to use the 't' option??
This is the current line in my extensions.conf
exten => 9998,1,Dial,SIP/9998
So would I change it to
exten => 9998,1,Dial,SIP/9998,t
Thanks.
- Original Message -
From: Pertti Pikkarainen <[EMAIL PROTECTED]>
Date: Fri, 14 Mar 2003 13:50:
Ok.. will try
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David T Hollis
Sent: 14 Maart 2003 14:10
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 segmentation faults
Liaan van der Merwe wrote:
> Ok.. you lost me..
>
> I'm new to this game.
>
Ok... will try this
But how do is tell asterisk to use this channel and not default..
???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lubomir Christov
Sent: 14 Maart 2003 13:45
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 segmentation faults
Liaan van der Merwe wrote:
Ok.. you lost me..
I’m new to this game.
Where can I check this.?
I install from cvs 2 days ago…
I had the same problem, though asterisk wouldn't even start. It would
bomb out when loading the chan_oh323 module. The solution was to not use
the CVS distro just yet,
Negative side effect with 't' option: all the local SIP-to-SIP media
streams travel trough Asterisk instead of going direct.
Right now I'm using SNOM's transfer option instead.
But now I can't use * call parking because of that. Using # is
probably better
if there are no scaling problems.
Re
use the System app,
like
exten => s,1,System(/path/to/your/application)
matteo.
> -Messaggio originale-
> Da: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Per conto di
> Christoph Schütz
> Inviato: giovedì 13 marzo 2003 22.46
> A: [EMAIL PROTECTED]
> Oggetto: [Asterisk-Users] Execute
Hello Liaan,
why you don't try oh323_channel ?
You can find it at:
http://www.inaccessnetworks.com/projects/asterisk-oh323
I think that it have better codec support (G723 )
Lubo
Liaan van der Merwe wrote:
Ok.. you lost me..
I’m new to this game.
Where can I check this.?
I install from cvs
why don't u switch dns off on the cisco ?
On Friday 14 March 2003 00:28, Lele Forzani shaped the electrons to say:
> On Thursday 13 March 2003 22:01, Mark Spencer wrote:
> > Can somebody look at the RTP packets with "ethereal" and tell me if they
> > notice any difference between what we send and
On 13 Mar 2003 at 13:00, Howard White wrote:
> To our multi-lingual listers - do IAX or TASTE have any non-English
> complications?
In German, "Taste" means key like in "keyboard", not "lock and key".
But the word is pronounced differently than the English "taste".
cu
Reinhard
_
Ok..
you lost me..
I’m
new to this game.
Where
can I check this.?
I
install from cvs 2 days ago…
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of Jeremy McNamara
Sent: 14 Maart 2003 11:19
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
If you search the archives you would find that for IP phone you need to
add a 't' option to the end of your dial command. The 't' option will
let the user dial '#' to get the systems attention, then dial an
extention for the transfer.
On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wro
Hi,
Firstly let me start off by saying that asterisk is one of the most amazing pieces of
open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and
even Linux itself.. Nice work!!
I have just installed my first server, thanks to the astinstall script.. and I have
read
Which H.323 channel driver you running? This debug doesn't look like
anyone i've seen
Jeremy McNamara
Liaan van der Merwe wrote:
Hallo all
When I try to make a call
to say 192 with h323 client (unkown extension) following logged.
Hallo all
When I try to make a call to say 192
with h323 client (unkown extension) following logged.
***
SetEndPointTypeInfo
1) OnIncomingCall from 192.168.1.108
1.1) After number init
2) gwcallToken is
ip$192.168.1.108:1336/
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