Dears
What's the major deference between Asterisk 1.6.0.6 and Asterisk 1.4.23
Regards
Khaled Chehab
NGN Eng.
Untitled
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail:
Dean Collins schrieb:
You want ADA which is the new name for the old snapanumber
I recently tried it and it is horrible. I could not check my bank
account because every time I clicked in Firefox on the bank account
number ada tried to call my bank account number. And when you send
emails
1.6.0.6
- 1.4.23
--
0.1.77.6 :-)
http://svn.digium.com/view/asterisk/branches/1.6.0/CHANGES?revision=172635view=co
klaus
Khaled W. Chehab schrieb:
Dears
What’s the major deference between Asterisk 1.6.0.6 and Asterisk 1.4.23
Regards**
*
Thanks,and kindly in which version of asterisk you advice to build a
business PBX ?
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: Thursday, March 05, 2009 12:41 PM
To: Asterisk
hi
in ael the [general] equivalent is globals { } ?
what is the equivalent in realtime???
can i use realtime and static ael or conf???
using realtime when i do sip show peers asterisk didnt show the realtime
peers what can i do?
Thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny
Hello all,
I have an extension like this in my test box:
exten = _,1,Answer()
exten = _,n,Playtones(350/100,0/750)
exten = _,n,Dial(SIP/${EXTEN})
The idea is that occasionally the SIP peer might take a few seconds to
connect the call, and this provides an in progress
Hi all,
I saw that there was an auto-provisioning feature on asterisk 1.6.x for
the Polycoms. But no real documentation.
I would like to know how, exactly, does the network has to be
configured to allow that. I used to provision my Polycom phones with the
help of tftp or ftp. But if there's a
Hmm that's a shame as Snapanumber which was the product/company that was
sold to Digium works fine for me.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
-Original Message-
Wednesday, March 4, 2009, 5:43:13 PM, Joseph wrote:
On 03/04/09 15:56, Gergo Csibra wrote:
Wednesday, March 4, 2009, 3:22:59 PM, Joseph wrote:
FAX Passthru Codec: G711u
for me FAX works better with G711a
Can you folks compare my setting below with your settings and let me know if
from where i can download ada?
David
2009/3/5 Dean Collins d...@cognation.net
Hmm that's a shame as Snapanumber which was the product/company that was
sold to Digium works fine for me.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642
Hi
I want to be able to use one provider if I dial 0 before the number and another
if I dial 1 before, how can I do that in asterisk 1.6?
/ralf
Ralf Träskman, IT
AdLibris AB, Box 3667, 103 59 Stockholm.
Besöksadress: Sveavägen 56C, 111 34, Stockholm - Obs ny address!
Dir: +46-(0)8-5460 60
ADA Forums: http://forums.digium.com/index.php?c=8
ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe
ADA Administrators Guide:
http://dl1.digium.com/ADA1.1/ADA_Admin_Manual.pdf
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357 New
Peter Mueller wrote:
Has anybody set up such an installation and/or is OpenAIS able to
transfer the devstates over different subnets? Haven't found docs and
hints for this use case.
The method OpenAIS uses to communicate between nodes is designed for a
very low latency local connection; it is
exten = _0X.,1,Dial(SIP/provider1/${EXTEN:1})
exten = _X.,1,Dial(SIP/provider2/${EXTEN:1})
2009/3/5 Ralf Träskman r...@adlibris.com
Hi
I want to be able to use one provider if I dial 0 before the number and
another if I dial 1 before, how can I do that in asterisk 1.6?
/ralf
exten = _0X.,1,Dial(SIP/provider1/${EXTEN:1})
exten = _X.,1,Dial(SIP/provider2/${EXTEN})
Sorry i copy paste and forgot to remove the :1 in the second line.
David
2009/3/5 David fire ddf...@gmail.com
exten = _0X.,1,Dial(SIP/provider1/${EXTEN:1})
exten = _X.,1,Dial(SIP/provider2/${EXTEN:1})
Wilton Helm wrote:
12kHz isn't really enough for high quality voice, and the extra bit
rate needed to push the bandwidth to 15kHz is small. Also, a deep man's
voice looses something when you cut off at 70Hz.
I'm not sure that this isn't stretching things a bit. There are no
handsets or
Problem is, without going to 1.6, I can't get the queue log or events
posted to MySQL in realtime.
There used to be a patch out there for queue_log, but it doesn't work
with versions 1.4.21 or higher.
--
Regards,
Robert Broyles
Anthony Francis wrote:
Robert Broyles wrote:
I saw some
I agree, the intersting part is adding what is not included in the standard
firmware.
Regarding documentation... On the one hand the phone is running a regular
embedded Linux, I think that does not require additional documentation. The API
to the phone is a different topic. It will really
Hi,
We are trying to use a softphone with instant messaging which uses
SIMPLE. i tried sending a message to the user, this is what i got:
[Mar 5 22:59:39] WARNING[1884]: chan_sip.c:9825 receive_message:
Received message to sip:1...@mydomain.com from
sip:3...@mydomain.com;tag=1d32305f,
Hello all,
Since a while some of our SIP users complain about gaps (sometimes
multiple seconds of silence) in the RX audio stream (direction pbx -
phones). Our configuration is an Asterisk with two Wildcard TE410P
cards that are connected with E1 PRI's to an external server running
On Thursday 05 March 2009 07:10:59 Kevin P. Fleming wrote:
Peter Mueller wrote:
Has anybody set up such an installation and/or is OpenAIS able to
transfer the devstates over different subnets? Haven't found docs and
hints for this use case.
The method OpenAIS uses to communicate between
The method OpenAIS uses to communicate between nodes is
designed for a
very low latency local connection; it is not designed to work across
routed connections. Russell Bryant has spent some time
talking to the
OpenAIS developers about this, but so far there doesn't seem to be a
good
Have someone running fine Alert-Info with a Snom 370
( System Information:
Phone Type: snom370-SIP
MAC-Address:0004132661BD
IP-Address: 192.168.10.170
Firmware-Version: snom370-SIP 7.3.14 14961)
i've tried
exten =
Steve Underwood wrote:
Good engineering of standards is about building them for the future.
Cutting off the bass at 70Hz is far less of a limitation than cutting
off the high end at 11kHz, but why do it in the codec? Why not leave the
transducers and their amps to do what they can?
Along
On Thu, Mar 5, 2009 at 5:09 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Along those lines, I've mentioned here before that my wideband Polycom
IP650 actually sounds substantially better with G.711 ulaw media streams
than any previous phone I've had, and I've had high quality phones
On Thu, 5 Mar 2009, Christian Stredicke wrote:
I agree, the intersting part is adding what is not included in the
standard firmware.
Regarding documentation... On the one hand the phone is running a
regular embedded Linux, I think that does not require additional
documentation. The API to
Hello,
I wonder if somebody can help me with following:
I need to acheive something similar to this:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
but with a twist.
Suppose i'm already in a meetme conf call
i want to dial a * for example, hear the dial tone
dial the destination
Kevin P. Fleming wrote:
Steve Underwood wrote:
Good engineering of standards is about building them for the future.
Cutting off the bass at 70Hz is far less of a limitation than cutting
off the high end at 11kHz, but why do it in the codec? Why not leave the
transducers and their amps
Steve Underwood wrote:
Kevin P. Fleming wrote:
Steve Underwood wrote:
Good engineering of standards is about building them for the future.
Cutting off the bass at 70Hz is far less of a limitation than cutting
off the high end at 11kHz, but why do it in the codec? Why not
Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a
patch way back for 1.2 that allowed all queue log events to sh,ow up in
the AMI, just haven't had time to make a new version for 1.6.
Maybe this time I can get the patch in trunk and it will always be there.
Robert Broyles
type in cli Core show application meetme and read how to use it
MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe
conference.
exten = 4105,n,meetme(99008664105|Ap)
So what conf number do you join here ? 99008664105
do you have a conf with that number ?
I have
The patch I was referring to is:
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging
It doesn't work for the current SVN 1.4
--
Regards,
Robert Broyles
Anthony Francis wrote:
Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a
patch way back
Hi all,
I'm getting the following error when receiving a FAX on Asterisk...
ERROR[11320]: chan_sip.c:12441 handle_response_invite: Got error on T.38
re-invite. Bad configuration. Peer needs to have T.38 disabled.
Any suggestions?
Regards
___
--
Adrià Vidal schrieb:
Have someone running fine Alert-Info with a Snom 370
exten =
200,1,SIPAddHeader(Alert-Info:http://www.notused.com\;info=alert-external)
exten = 200,n,Dial(SIP/${EXTEN},30)
Can see into the phone SIP trace is receivend the Alert-Info, but
phone continue to ring with
Hmm, Yeah something that writes direct to MySQL would not have made it
into trunk, I took the route of using the queue event flag to also turn
on and off sending of all queue events normally written to the log to
also go to the AMI. I also have a perl script that listens to the AMI
for events
For those who are using SuSE:
At last they've managed to create ready-to-run packages for
openSUSE_11.1. They are there since a couple of hours...
(For other versions it was allready available for some time on the OBS)
Is there a built-in way of detecting fax tones, or a switch to T.38 on a
SIP channel? I need to periodically check some efax servers for
availability and figured the best way to ensure they are operational is
to check for tones. I've looked into Nvdetect but the company seems to
have gone out
Christian Stredicke wrote:
I agree, the interesting part is adding what is not included in the
standard firmware.
Regarding documentation... On the one hand the phone is running a
regular embedded Linux, I think that does not require additional
documentation. The API to the phone is a
Thank you Gry,I think i found the reseaon:maybe I shoule re-install the
zaptel.I have reinstall the zaptel and it worked well.
Thank you ,thank you for your help,Dear Gry!
regard
Qiu
2009-03-06
邱磊
发件人： Grygoriy Dobrovolskyy
发送时间： 2009-03-06 02:42:51
收件人： Asterisk Users Mailing
Here is the current dial plan section:
[custom-michael]
exten = _900,1,Playback(custom/extn-xfer)
exten = _900,2,SayDigits(${EXTEN})
exten = _900,3,MixMonitor...
exten = _900,4,Dial(SIP/${EXTEN}|${DEFRT})
exten = _900,5,Playback(custom/extn-xfer2)
exten = _900,6,Goto(custom-michael,901,4)
Hi all,
Occasionally, DIDs from different providers stop working for some reason.
I would like to be able to monitor situations like that and react before any of
my clients start going ballistic on me.
Any ideas? Scripts you know of or wrote and willing to share?
Any info would be greatly
Have a look for agx-ast-addons and spandsp.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert McGilvray
Sent: 06 March 2009 01:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fax
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Michael mich...@networkstuff.co.nz
Reply-To: mich...@networkstuff.co.nz, Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Date: Fri, 6 Mar 2009 15:52:20 +1300
To: Asterisk
hello,
I will do a server to do a lots of conferences (MeetMe).
I want to know that if I dont use a digum card, the limit of simultaneous
calls is harder without a card than with a card ?if, yes, how harder is the
limit?
thank you
Cordialement,
BERGANZ François
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