Hi
I am running firewall/router brew made of RedHat, Shorewall, Siproxd and
GnuGK on a box that connects through PPPoE to Internet. I run Asterisk on
another box behind of it and it seem to work fine for me.
I am thinking of replacing the router box because hardware is getting flaky.
I do not
* console just prints:
-- Hungup 'Zap/1-1
Thank you
Serge
From: James Golovich [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problem with Manager Originate
Date: Sun, 4 Apr 2004 15:02:42 -0400 (EDT)
On Sun, 4 Apr 2004, Serge Mankovski
HI
If I understand correctly, you are talking about a production of RSS feed
(see http://www.xml.com/lpt/a/2002/12/18/dive-into-xml.html)
I am writing a bunch of java classes that will expose Manager interface in
more readable form (form Java point of view). I might think of writing an
RSS
Hi
I am trying Manager interface for originate a call. This is what I get
---
Action: Originate
Exten: 555
CallerID: test 6656
Context: local
Timeout: 600
Channel: SIP/8782
Priority: 1
Response: Error
Message: Originate failed
What do I do wrong?
Thank you
Serge
Is it possible to have ActionID on manager interface messages if call is
originated from a .call file?
From: Nicolas Bougues [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Identifying a call with manager interface
Date: Fri, 19 Mar 2004
, 2004-03-16 at 17:44, Serge Mankovski wrote:
Hi
I have written a web service that starts a conference call and then
monitors
call progress on the manager interface. It works nicely until conference
in
a voice mail system. It would be better if I could intercept the fact
that
the answering side
and voice
mail
Date: Tue, 16 Mar 2004 11:05:32 -0600
is it publicly available?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
Serge Mankovski
Sent: Tuesday, March 16, 2004 10:45 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Web service to start
Hi
Here is my problem: I initiate a conference call by placing several .call
files into /var/spool/asterisk/outgoing/ directory
Asterisk starts calls and I can see events in the manager interface.
At the same times there are other calls going on and there are many more
events in the manager
Hi
Here is my problem: I initiate a conference call by placing several .call
files into /var/spool/asterisk/outgoing/ directory
Asterisk starts calls and I can see events in the manager interface.
At the same times there are other calls going on and there are many more
events in the manager
_
Add photos to your e-mail with MSN 8. Get 2 months FREE*.
http://join.msn.com/?page=features/photospgmarket=en-caRU=http%3a%2f%2fjoin.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca
Hi
I am trying to dial an extention on my gateway using OH323 without a
gatekeeper.
I would like to be able to do this:
exten=_8.,1Dial(OH323/($EXTEN:1)@xxx.xxx.xxx.xxx,20,r)
It seems that the only way I can dial via OH323 is
exten=_8.,1Dial(OH323/xxx.xxx.xxx.xxx,20,r)
Any incite into diling
Hi,
r in Dial statement in extentions.conf has to provide ringing tone to the
calling party. It is described in the documentation.
However, I experience similar behavior (no ringing tone) although I have
Dial(SIP/whatever,20,r) in extentions.conf
This happens every time when there is a call
Hi
I am using DISA to get my Polycom SoundPoint400 with H323 firmware to
connect to *
I have it working, but when I dial SIP end points there is no ringing tone
on the phone. DISA gives dial tone but does not give ringing (if I
understand correctly it is because it expects to transmit sound
21:01:53 +0200
Citeren Serge Mankovski [EMAIL PROTECTED]:
Hi
what can be wrong with * that console does not show any stderr text
printed
from agi script?
I am starting with asterisk -rc
VERBOSE command does show text on console but printing of STDERR does
not
If you
Hi All
I am having this problem when setting up a H323 call.
Can anybody tell me what is going on?
Thanks
Serge
--
NOTICE[245776]: File chan_oh323.c, Line 1293 (oh323_write): H323:1637:
Format changed from 4 to 8.
WARNING[245776]: File frame.c, Line 76 (ast_smoother_feed):
Hi
what can be wrong with * that console does not show any stderr text printed
from agi script?
I am starting with asterisk -rc
VERBOSE command does show text on console but printing of STDERR does not
I tried it from Perl and from Java and in both cases almost the same
at the top of the Perl script:
$| = 1;
select((select(STDERR),$| = 1)[0]);
This removes buffering.
quote who=Serge Mankovski
Hi
what can be wrong with * that console does not show any stderr text
printed
from agi script?
I am starting with asterisk -rc
VERBOSE command does
Hi All,
I know, IAX is NAT friendly, but... I have a problem running gnophone from a
box behind NAT firewall.
I can register gnophone with * through NAT, but when I try to make a call it
instantly disconnects. CLI
iax show peers command tells me that peer is unreachable. However this peer
is
in this case.
-wade
-Original Message-
From: Serge Mankovski [mailto:[EMAIL PROTECTED]
Sent: Monday, September 08, 2003 6:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Help needed with IAX behind NAT
Hi All,
I know, IAX is NAT friendly, but... I have a problem running
Hi
Did anyone try to setup DLink DG-104S VoIP Gateway with asterisk?
Thanks
Serge
_
MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*.
http://join.msn.com/?page=features/virus
___
Hi All,
i found an article that explains SIP NAT woes.
http://www.sipcenter.com/files/SIPNATtraversal.pdf
It is a great read for all people in the mailing list that have problems
with SIP when * is behind NAT or when there is NAT between asterisk and a
SIP phone.
Serge
Hi,
I updated kernel for RH8.0 and updated * from cvs. After that zaptel compile
exits with error. Is it because of new kernel or zaptel source code change?
In any case could somebody help me to fix this problem?
Thanks,
Serge
the README.. Please read that over again.. it
tells you exactly what to do.
bkw
On Sun, 10 Aug 2003, Serge Mankovski wrote:
Hi
I am using inAccess channel driver.
Compiled, installed. This is what I get when I am trying to start *
---
[chan_oh323.so]WARNING[16384
Hi
I am using inAccess channel driver.
Compiled, installed. This is what I get when I am trying to start *
---
[chan_oh323.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource):
libh323_linux_x86_r.so.1.12: cannot open shared object file: No such file
Hi
I am trying to call from Linphone on extention 109 to Xlite on extention 108
and I get this error
--
to 216.75.167.18:5068
WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application
'Dial ' for extension (sip, 108, 1)
== Spawn extension (sip, 108, 1)
Hi
I want to set up * as a conference bridge. I would like to be able to
conference is SIP calls (up to 12)
I am looking through all available documentation for * to get info on how it
is done. No luck so far.
Can somebody direct me to the info in this subject?
Thank you
Serge
: Re: [Asterisk-Users] Compiling Asterisk under Yellow Dog
Date: 24 Jun 2003 18:28:51 -0500
On Tue, 2003-06-24 at 17:59, Serge Mankovski wrote:
Hi,
I am trying to compile Asterisk under Yellow Dog 3.0 distributionn.
I am getting an error
gcc -shared -Xlinker -x -o codec_gsm.so codec_gsm.o -lgsm
27 matches
Mail list logo
