OK. I tried with 1.12.2 and indeed problem fixed. Thanks, Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Sbado, 13 de Diciembre de 2003 04:54 a.m.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RH9 and h323.conf
SW
On Fri, 2003-12-12 at 09:29, Dan wrote:
Hi,
- Original Message -
From: Alastair Maw [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 4:58 PM
Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
On 12/12/03 13:56, Dan wrote:
This
Hi folks,
To provide MWI, * will send out a sip:notify message to the UA.
The originator of this message is asterisk, as shown below;
NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0
Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21
From: asterisk sip:[EMAIL PROTECTED];tag=as0ffb1bdc
I tried again at runlevel 3 but to no avail.
I'm pretty sure I have sufficient horsepower since I'm running on a box
with
half gig memory and a speedy CPU.
burak
I run on a Pentium I /100 Mhz, 32MB RAM with RedHat 9.0 and have no
trouble with voicemail audio or Music On Hold. This is a
SATURDAY 20th
I have had far fewer emails than the noise created earlier about Mark's
arrival in Paris. Everyone who has contacted me I have replied to once.
Again please - if you want to come please email me at [EMAIL PROTECTED] I will
set a venue and time tomorrow evening to email to all
Today I deleted the files in the asterisk, libpri, zaptel directories that
are in /usr/src and did a new CVS checkout (not update). After doing
the make installs and starting asterisk the show version is the same
as before:
Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586
Hi,
From: rnc Info Lists [EMAIL PROTECTED]
Today I deleted the files in the asterisk, libpri, zaptel directories that
are in /usr/src and did a new CVS checkout (not update). After doing
the make installs and starting asterisk the show version is the same
as before:
Asterisk
On Sat, 2003-12-13 at 05:14, Dan wrote:
Hi,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
Sounds like you want app_rcfax at the pstn to decode the fax data, then
it would somehow transfer the image file to your local machine via the
slow link and use
I just updated yesterday, but I did a complete rm -Rf for all of the
following directories:
/usr/src/zaptel
/usr/src/zapata
/usr/src/libpri
/usr/src/asterisk
Then I did a new cvs checkout for all four of those items before
recompiling them (make clean; make install) in the
On Friday 12 December 2003 22:04, Ulexus wrote:
On Thursday, 13 November, 2003 11:34, Tilghman Lesher wrote:
On Thursday 13 November 2003 07:31, Marcus Adolfsson wrote:
How does a user exit the directory application?
Say he can't find the person that he is looking for and wants to
Did you do a make clean first before the make make install ?
Just my $.02 :)
-bh
Quoting rnc Info Lists [EMAIL PROTECTED]:
Today I deleted the files in the asterisk, libpri, zaptel directories that
are in /usr/src and did a new CVS checkout (not update). After doing
the make installs
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote:
I just updated yesterday, but I did a complete rm -Rf for all of the
following directories:
/usr/src/zaptel
/usr/src/zapata
/usr/src/libpri
/usr/src/asterisk
Then I did a new cvs checkout for all four of those items
Hi,
From: Steven Critchfield [EMAIL PROTECTED]
While I haven't tested it much, my first use of app_rxfax and app_txfax
worked fine. The glue is little more than procmail and sample.call
combined with working mail servers.
I have faith that Steve's work will continue to make the functionality
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote:
I just updated yesterday, but I did a complete rm -Rf for all of the
following directories:
/usr/src/zaptel
/usr/src/zapata
/usr/src/libpri
/usr/src/asterisk
Then I did a new cvs checkout for all four of those items
Burak,
Try connecting to your * server with a SIP phone like X-Lite or an IAX
phone like DIAX. Do you get the same results with those phones too?
Jonathan
Burak Balasaygun wrote:
Hi,
I just got started with asterisk and am having a problem with voice quality.
When connecting via either a
Hi!
The line with ;sock=/tmp/mysql.sock, i think you must write it without the ;.
You need this socket to connect with mysql.
Best regards,
Mireia
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of listas
iPfone
Sent: sexta-feira, 12 de dezembro de
On Saturday 13 December 2003 11:02, Mireia Munoz de jesus wrote:
The line with ;sock=/tmp/mysql.sock, i think you must write it
without the ;. You need this socket to connect with mysql.
You don't usually need that configuration line. It's only there if your
server and client conflict about
Some people on this group may have understood from messages posted
here that the Sipura SPA-2000 is not currently available for shipping. That is
not the case. Voxilla.com has the Sipura SPA-2000 available for immediate
shipping, and has had them since late November. The price is $109.95,
Jonathan,
I have it sorted it out. Rebooted the box and it works fine on all
interfaces now.
thanx
burak
On Sat, 13 Dec 2003 08:40:26 -0800, Jonathan Tew wrote
Burak,
Try connecting to your * server with a SIP phone like X-Lite or an
IAX phone like DIAX. Do you get the same results
I'm using a modified default config file for extensions.conf, the one
that uses macro-stdexten to handle the stations.
We use a TDM30 card for our stations.
When a call that has been rung in using that macro transfers the call
things work just fine as far as the other instrument ringing.
But
Their is an open bug on this on bugs.digium.com it doesn't happen on #
transfers just flash transfers from what I can tell.
bkw
On Sat, 13 Dec 2003, Brian Capouch wrote:
I'm using a modified default config file for extensions.conf, the one
that uses macro-stdexten to handle the stations.
We
Timothy Costello wrote:
and somewhere (maybe on the wiki) should be a link to ESR's How to Ask
Smart Questions: http://www.catb.org/~esr/faqs/smart-questions.html
I know it's been posted to the list several times. It should be part of
the FAQ to read it before asking questions...
Added link
Hi!
When a call that has been rung in using that macro transfers the call
things work just fine as far as the other instrument ringing.
But once the ring timeout has expired, the call then drops into the
*original station's* voicemail.
This is a known bug:
Tilghman Lesher wrote:
On Friday 12 December 2003 07:25, Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer fax
data (even converted in another format) between Asterisk boxes,
using low bandwidth codecs like GSM.
I know that this is possible only with the G.711 now
I'm having problems retrieving messages.
1 I dial the ext to run VoiceMailMain
2 VoiceMailMain asks for my password
3 I enter password (via BT-100 phone) on the keypad followed by #.
4. When examining the asterisk console I see some of the digits I entered for
my passwd are repeated. For
When one exension has a voicemail, stutter dialtone is turned-on for all
extensions.
My zapata.conf entries:
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
My voicemail.conf entries:
[other]
61 = 1234,...,[EMAIL PROTECTED]
62 = 1234,...,[EMAIL PROTECTED]
etc.
Are my
Incorrect Password '4433211' for user '2000' (context =any)
This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream
Note: Don't forget to reload after modifying sip.conf.
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL
Can anyone point me in the right direction?
I'm trying to get asterisk to use my ALSA compatible sound card(configured
correctly) and my Conexant modem to get on an outside line.
Or can someone point me to where the documenation for teh ALSA conf is
located?
I felt competent wiht linux, but
_
Add photos to your e-mail with MSN 8. Get 2 months FREE*.
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Thanks a lot. It works fine now
On Sat, 13 Dec 2003 21:27:21 +0100, Philipp von Klitzing wrote
Incorrect Password '4433211' for user '2000' (context =any)
This is a FAQ: use dtmfmode=info in your sip.conf for your
Grandstream Note: Don't forget to reload after modifying sip.conf.
On Sat, 2003-12-13 at 10:32, rnc Info Lists wrote:
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote:
I just updated yesterday, but I did a complete rm -Rf for all of the
following directories:
/usr/src/zaptel
/usr/src/zapata
/usr/src/libpri
/usr/src/asterisk
I don't get why people always say dtmfmode=info mine works fine with
rfc2833.
bkw
On Sat, 13 Dec 2003, Philipp von Klitzing wrote:
Incorrect Password '4433211' for user '2000' (context =any)
This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream
Note: Don't forget to reload
When asterisks gets and incoming call the server is setup to dial a zap
phone and then plays a message telling the caller what extension to
press to leave a message. During that message I can not press the
voicemail extension or any other extension that is included.
Thank in advance for any
I have a problem with IAX call transfer. The call goes successful but
consumes lot of BW in the middle tier.
The actual connection is like this
(NAT) DIAX(IAX2) - *1 -- *2
*1 *2 were public IP with asterisk.
It consumes around 120kbps in total to forward a single GSM call.
I have the
Hi!
I don't get why people always say dtmfmode=info mine works fine with
rfc2833.
bkw
Dunno. I tried rfc2833 first, and had exactly the same problem as
described below with voicemail (but only there). Info then worked just
fine (as obviously also confirmed by this user here).
Is there any
On Sat, 2003-12-13 at 22:18, Matthew Pallotta wrote:
When asterisks gets and incoming call the server is setup to dial a zap
phone and then plays a message telling the caller what extension to
press to leave a message. During that message I can not press the
voicemail extension or any other
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