the CDR. So, a simple NoCDR() before dialing should work, or ForkCDR()
and then NoCDR() if you want to save previous data.
Regards
On 8/27/06, Master Abi [EMAIL PROTECTED] wrote:
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just
That is what I thought, but then how do I STOP recording CDR's. If I use
it in the h extension, it also gives a warning.
Moises Silva wrote:
Normal behaviour since the call record before executing NoCDR() was
not posted (saved)
Regards
On 8/28/06, Master Abi [EMAIL PROTECTED] wrote:
When I
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just not working for me. My dial line is
exten = _70XX,1,Dial(SIP/${EXTEN}|20|Ctr)
Could someone give me a short example of using NoCDR correctly.
Thanks
Master
Hi
I am about the purchase a server and would like to know if anyone has
had any experience with the TE410P Rev 2 in a server that has a
ServerWorks BCM5785 (HT-1000) chipset.
Thanks
Master
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Cory,
An easier way to do it: (used gentoo)
1. Connect a PATA drive and install gentoo with 2.6.14 and include
Marvell SATA driver.
2. Use Ghost for Linux V.0.17 and copy PATA disk to SATA disk.
3. Disconnect the PATA
4. Boot from the install CD and change grub.conf and fstab
5. Reboot
Hi
Upgraded to Gentoo 2.6.14-r2. When compiling zaptel, warning appears.
Zaptel module loads fine.
Cannot remember seeing this on 2.6.13. Is there another Kernel switch
that needs to set. CRC and RTC is set in kernel.
make[1]: Entering directory `/usr/src/linux-2.6.14-gentoo-r2'
CC [M]
Hi
I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after
putting in the upgraded board but did not change any conf, but the spans
become active but will not come up.
I guess I am missing something or are the any changes to the
zaptel/libpri software that is required. I
make; make install i also executed make config. This copies the
correct startup script to /etc/init.d/zaptel. Without this it also
didn't worked for me.
Master Abi wrote:
Hi
I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after
putting in the upgraded board but did not change
Hi
I am trying to develop a night divert. Caller dials in after hours on
Zap and it gets divert to a mobile number via a second Zap. The call
bridges but will not hangup the channels when the parties finish.
Is there something I am missing or an dial option that I should be
using. I am
Hi all
Could someone please care to share an example of the Dial W option
usage. I cannot seem to find any reference to it usage. I know you use
*1 in features.conf to start the monitor, but from there I am lost.
Master
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if I have conf = 80,111 in meetme.conf, I dial 80# and connect to the
conference, then I dial 111#, it indicates pin is incorrect. with other
phones it works. Is there something special in the sipura config that
will allow more digits after the #
master
Craig wrote:
I found the speaker phone
I use the MII 1.2Ghz version with TE110P. No problems. Can do about 8-10
ulaw to GSM, possibly more. Also used TDM400 that works fine. Note the
MII 1.2 version cannot boot off the CF unless you use FreeBios. Use the
EPIA MS version to boot from onboard CF.
C. Tomlinson wrote:
Hi,
I run * on
any backlit on the display. So I think that not.
On Sun, 13 Mar 2005 23:31:03 +1100, Master Abi [EMAIL PROTECTED] wrote:
Hi
Just 2 issues I have with SPA841.
1. I autodial extension 600 then inside an AGI wait for more digits.
The digits are transmitted correctly to * but they do not show up
Hi
Just 2 issues I have with SPA841.
1. I autodial extension 600 then inside an AGI wait for more digits.
The digits are transmitted correctly to * but they do not show up on the
SPA841 display, only the 600. How do I set the 841 is show the digits
after the 600#
2. Is the SPA841 pixel
Could you email me the PDF I am having PASV FTp problems. I have the
same setup. Out of interest which case are you using. I looked at the CF
adaptor you used, but not sure if the Morex 3677 case I am using is high
enough.
Kilburn
JR Richardson wrote:
Hi all,
The journey is complete, at
Hi,
G726-32 codec from beta firmware 1.0.4.54 now works fine with *. Tested
on BT101 and HT286 over a 64K DSL line. Some progress but iLBC still has
not surfaced.
Get it from http://www.grandstream.com/BETATEST/
Master
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Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works
great.
Master
Greg Boehnlein wrote:
Hello all,
I'm trying to get the g726 codec patch contained in:
http://bugs.digium.com/bug_view_page.php?bug_id=0001104 to
I am not running the V1-0stable. Use the development version. My version
is 2 days old. G726 added to development CVS about 10 days ago.
Greg Boehnlein wrote:
On Mon, 8 Mar 2004, Master Abi wrote:
Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy
Hi
What is the relationship between when CDR recording occurs and the
hangup extension is executed. Normally CDR happens before the h
extension is executed.
I use the h extension to clean up for routines, but sometimes it gets
called to quickly before the CDR is dumped into a DB. I would like
Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium
Zaptel cards?
Matteo Brancaleoni wrote:
Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto:
Klaus-Peter Junghanns [EMAIL PROTECTED] said:
we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
One thing I'd
I think this is related to a device (GS in my case) that has an sip
entry but you physically removed it and switched it off. Somehow * still
thinks connected. Comment out the entry and reload or put the device back.
Mark Rizzo wrote:
I have seen similar error which coincided with my GS phone
Checked the archives. I cannot get ADPCM to work with SIP. Calling from
phone1 (adpcm) to phone 2(ulaw). Both phones Grandstreams with one set
with G726-32 with v0.7.1 cvs. Has anyone got adpcm to work?
Jan 24 09:00:14 WARNING[409617]: rtp.c:1069 ast_rtp_write: Not sure
about sending format
Aastra will have a production PT480i SIP phone in March for ~US180-$200.
Same phone as ADSI model just SIP, but has 4 extra buttons for virtual
lines. Got a beta SIP model under test. Designed for SIP v1 v2. * is
one of PBX used for testing by development, so should be * friendly when
I had experienced this problem before. I found this to be related to 2
items. Firstly, try not to use the s,1 starting each submenu. Secondly,
if there are more than 20 sub menus, you will get this delay problem.
Why I do not know. I reordered and regrouped and the problem
disappeared.
PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, 13 November 2003 2:11 PM
To: Master Abi
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] pause after dialed option
So what do you use instead of s,1? My s extensions set things like
response timeout, digit timeout, etc. Thanks again.
AJ
10 - A way to lock the phone settings (IP address, etc). It is too easy
to change the settings when in a public environment. The MENU button
should not be 1 press away from changing the settings, Use MENU + SOME
COMBINATION.
7 - Use the conference button to access Meetme. Like the Voice Mail
Hi,
After I hear the intro, I press 1 or 2 and I get a delay of about 5
seconds before the 1 or 2 exten is read. I am sure this worked without a
delay before. I did a CVS upg about a week ago.
I also just tried it with a single background statement, same result.
Could be related to the
I filed a bug report yesterday about it.
http://bugs.digium.com/bug_view_page.php?bug_id=330
Budgetones are effected, not sure about others. It seems to be codec
related. If you use allow=all, then it tries to negotiate G723 with Ulaw
and this effects other audio items.
MA
-Original
Title: Message
Hi,
Checked out latest
CVS and no sound from Playback, Background, MOHor bridged
channels.mpg123 is active but no sound.
Master
Hi,
Checked out latest CVS and no sound from Playback, Background, MOH or
bridged channels. mpg123 is active but no sound.
Master
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Title: Message
Hi
I read through the archives but could not find much
reference to * using SIP on TCP instead of UDP for signalling. Can * be
configuredand if so how. My service provider will only accept SIP
signalling on TCP.
Thanks
Master
JT,
We use 2 providers iPCB.NET and NTT (backup) and both require signalling
on TCP only. Interestingly, I find this to be the norm amongst Cisco
powered providers.
As * marches on to the #1 telco product and SIP to the #1 protocol of
choice, protocol=[tcp,udp,auto] feature is a good idea in
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