Hi,
Tomorrow we'll be talking to some of the guys at
http://www.safisystems.com/ about their visual call flow and IVR
software. Also more about asterisk 1.6 changes and (we hope)
improvements, wide-band audio, and possibly the betas for X-Pro for
Asterisk and Skype channel modules.
You get there
Hi,
Does anyone have a suggestion how I can analyze the concurrent usage of
ISDN channels? A client complains about their clients sometimes getting
a busy tone when trying to call them. I suspect they just need to add an
additional ISDN2 line but I need some conclusive information that they
Dear all,
Does asterisk supports H323?If yes how to enable it?
Regards
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Has anyone done a modification where you can Interrupt Asterisk's
SayDigits(). This will be helpful in order to be able to interrupt an
announce and dial digits without waiting to hear all the announcements.
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Yes, this has already been answered. Search previous post for
implementation.
On Thu, Oct 9, 2008 at 3:34 AM, michel freiha [EMAIL PROTECTED] wrote:
Dear all,
Does asterisk supports H323?If yes how to enable it?
Regards
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Hello Robert and all others of course! :-)
Here are the ringtones (ring files):
http://juliencoder.de/ringtones.tar.bz2
All are in .wav-format, once in CD-quality and once in phone-line-quality.
All means all 6 of them. If the console/dsp only plays gsm or some other
special format, which
Patrick schrieb:
Hi,
Does anyone have a suggestion how I can analyze the concurrent usage of
ISDN channels? A client complains about their clients sometimes getting
a busy tone when trying to call them. I suspect they just need to add an
additional ISDN2 line but I need some conclusive
Hi!
One further notice about ringtone6: This existed long before today. It's
called schon wie-der drei-zehn To-te, the dashes are there to mark
syllables. The translation is: Again 13 dead people. the the melody of a
German radiostation's traffic news. A German commedian came up with the
Hi,
if possible use 7906G without callmanager software but only with SIP
protocol support ?
Thanks.
--
Salvatore.
- Original Message -
From: Stefan Gofferje [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Mike,
Can you tell us :
- asterisk version
- zaptel version
When you call over this line, when you hangup did you hear an busy
tone ? or any class tone ? To do this test connect your lines to
analog phone and make a call. Let's us know the results.
Regards,
Luis Morales
On Fri, Oct 10, 2008
Thank you very much.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julien Claassen
Sent: Thursday, October 09, 2008 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ringtones for the console
Hi Dave,
the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside
has:
apps11.1-1-3-15.sbn
cnu11.3-1-3-15.sbn
copstart.sh
cvm11sip.8-0-3-16.sbn
dsp11.1-1-3-15.sbn
jar11sip.8-0-3-16.sbn
load307
load369
SIP11.8-0-4SR1S.loads
term06.default.loads
term11.default.loads
I use
On Thu, Oct 9, 2008 at 6:38 AM, C. Savinovich
[EMAIL PROTECTED] wrote:
I am tinkering with a new router, a Linksys wrt610n dual-band, etc. But
the when I connect it, the softphones(x-lite) on the computers don't even
register. After a couple of hours of hassle, I found out that if I dmz the
Sasa,
Sometimes I have to do a hard reset of the phone in order to get it to load the
SIP firmware, even when the load file is specified in the SEPMAC.conf file.
Make sure that only the contents of the cop file and the SEPmac.cnf file are
present in your tftp root. Then unplug the phone and
Sasa,
You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't
have (to my knowledge) any non-callmanager SIP software. The SIP load is just a
SIP load, not a SIP load unique to generic SIP or callmanager.
Dave
-Original Message-
From: [EMAIL PROTECTED]
Hi I have searched the mailing lists and come across similar threads,
but no actual solution. I am trying to use a Cisco AS5300 as a
gateway for PSTNr. I have been able to configure it to take outbound
calls and send them to the PSTN just fine. Inbound calls however are
rejected by
What did the firewall change from and to? Did you have NAT enabled in *
AND on the Cisco phones? FYI, if you have NAT enabled in both places,
it will work if you have NAT in your setup or not. If you don't have it
enabled in both places, then it may or may not work depending on your setup.
As a followup to my previous email, change nat_enable to 1 and reboot
the phones.
Jerry Geis wrote:
Did you check sip.conf to make sure that the port is correctly set to 5060?
Please show the output of Cli sip show peer peernumber and the contents
of your SEPMAC.cnf file.
Dave
This is due to an SDP mismatch of some sort, codec or otherwise.
Perhaps you have not set your Asterisk SIP peers to support RFC2833
DTMF? Try dtmfmode=rfc2833. Either that, or your Asterisk SIP peers
are not accepting the gateway's offer of G.711u.
Of course, I have seen interop bugs in
On Wednesday 08 October 2008 22:05:14 Rob Hillis wrote:
Tilghman Lesher wrote:
On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
Wesley Haut wrote:
Yell at me if you will, but I hate func_realtime - it's not very
usable nor is it change-friendly (update your database and your
dtmf mode was set in the sip.conf
dtmfmode=rfc2833
I will remove the other codecs from sip.conf and see what effect it
has. Do you see any other potential issues in the configs?
thanks
On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote:
This is due to an SDP mismatch of some sort, codec or
Not offhand / without seeing the Asterisk side.
On Thu, October 9, 2008 10:26 am, Ketema Harris wrote:
dtmf mode was set in the sip.conf
dtmfmode=rfc2833
I will remove the other codecs from sip.conf and see what effect it
has. Do you see any other potential issues in the configs?
thanks
On Oct 8, 2008, at 11:41 PM, Carlos Chavez wrote:
I have a customer that wants to use meetme but they want to have
the users
record their name so it is played to the other people on the
conference. Is
there an easy way to do this?
--
Carlos Chavez
Director de Tecnología
Hi I have a Panasonic TDA600 conected with one E1 to the pstn and one PRI with
my Asterisk using a Digium TE220B card.
The Panasonic is master clock and Asterisk is slave, additionally the
Panasonic takes the clock from the PSTN E1. This scheme works and I´m able to
make and receive calls
Hi Dave,
I have tried restore to factory default value (as you have recommended to
me) but without success, however also with only files:
SEPMAC.conf file
contents of the cop file
..but the result isn't changed !
Thanks in advance.
--
Salvatore.
- Original Message -
From:
Please send the TFTP log after using the regular factory reset method I
described.
Thanks
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
Sent: Thursday, October 09, 2008 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi, sorry for my insistence but for me is a big problem ! :-( ...someone
have the same problem ?
Thanks in advance.
--
Salvatore.
- Original Message -
From: Sasa [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Try with fop,
http://www.asternic.org/
Regards,
Luis Morales
On Fri, Oct 10, 2008 at 2:40 AM, Patrick
[EMAIL PROTECTED] wrote:
Hi,
Does anyone have a suggestion how I can analyze the concurrent usage of
ISDN channels? A client complains about their clients sometimes getting
a busy tone
Folks,
I've seen a few reports that people have had problems with hang up
detection on UK cable phone lines. I have a TDM400P with two FXO ports,
one connected to my BT line and the other connected to my
Telewest/Virgin Media cable line. If I ring the BT line and then clear
down, Asterisk
Using the Community Edition
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
--
Date: Thu, 9 Oct 2008 00:37:34 +0200
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re:
Sasa schrieb:
I need other files other than those obtained with
cmterm-7911_7906-sip.8-0-4sr1.cop ??
cmterm is the callmanager software. You need to get the non-callmanager
SIP-software. Contact your local Cisco representative to buy a license
for that.
Terve,
Stefan
--
Last words of a
Hello,
I have an intermittent problem whereby the Asterisk (1.2.30.1 on fc9)
process unexpectedly stops.
It seems to occur when a call is being parked.
Below is an output from the full log, but as you can see it's not throwing
much light on the problem.
Any help with this is greatly
First off thank you for your help, using your help in conjunction with a
couple of my own changes it partially worked. I got rid of the iax-incoming
context, it seemed useless. I may be wrong in that assumption.
Looking back at what I have now:
Extensions.conf on server A
[vvfarm-extensions]
Jerry Geis wrote:
Did you check sip.conf to make sure that the port is correctly set to
5060?
Please show the output of Cli sip show peer peernumber and the
contents of your SEPMAC.cnf file.
Dave
This all ended up being CRAZY network stuff.
my server has 2 network cards in it.
On 10/9/08, Ketema Harris [EMAIL PROTECTED] wrote:
Hi I have searched the mailing lists and come across similar threads, but no
actual solution. I am trying to use a Cisco AS5300 as a gateway for PSTNr.
I have been able to configure it to take outbound calls and send them to the
PSTN just
On Oct 9, 2008, at 10:40 AM, [EMAIL PROTECTED]
wrote:
Hi I have searched the mailing lists and come across similar threads,
but no actual solution. I am trying to use a Cisco AS5300 as a
gateway for PSTNr. I have been able to configure it to take outbound
calls and send them to the PSTN
On Thu, 9 Oct 2008, Mike wrote:
Folks,
I've seen a few reports that people have had problems with hang up
detection on UK cable phone lines. I have a TDM400P with two FXO ports,
one connected to my BT line and the other connected to my
Telewest/Virgin Media cable line. If I ring the BT
On 10:21, Thu 09 Oct 08, Steve Anness wrote:
First off thank you for your help, using your help in conjunction with a
couple of my own changes it partially worked. I got rid of the iax-incoming
context, it seemed useless. I may be wrong in that assumption.
Looking back at what I have now:
Ok!!
Do this finale test.
call fron analog lines to any number, wait until the called hang up.
Now tell us the signal tone. If the signal is busy ? or any thing
else
Your set up look ok. Now try with this options into zapata.conf header:
zapata.conf:
usecallerid=yes
hidecallerid=no
I have a weird problem with a client. I recently upgraded to Asterisk
1.4.22 and Zaptel 1.4.12.1 on their server and now there is a problem
when a fax call is received.
Basically when faxdetect=incoming is set in zapata.conf the call comes
in and the fax extension dials a Linksys
Quoth Jared Smith...
On Wed, 2008-10-08 at 18:51 +0100, Russell Brown wrote:
Can anyone point me at a code fragment (C would be nice) that I could
use to subscribe to hints on a * box?
I'd like to write a small (hopefuly efficient) widget to show custom
device states and believe that a
On Fri, 10 Oct 2008, Luis Morales wrote:
Ok!!
Do this finale test.
Who, me or the OP (Mike). My setup works OK and I've no intention of doing
tests, final or otherwise, thanks.
Gordon
call fron analog lines to any number, wait until the called hang up.
Now tell us the signal tone. If
I have a TDM400 working quite well, Digium dialled in and recompiled
chan_zap with some changes , to get BT Callerid working and I have set
hangup on polarity in the zaptel.conf which seems to work well
this is a BT home line, not business, if you have a business line you
should get the
Thanks for the all the help, I have been pulling my hair out
I now have the trunk working in both directions. However, how do I add
voicemail capability?
exten = _11XXX,1,Dial(iax2/colo/${EXTEN:2},20,Ttr)
exten = _11XXX,n,Voicemail(${EXTEN:2:3}|su)
Thinking that if I dialed 11127 and after 20
Russell Brown schrieb:
Quoth Jared Smith...
On Wed, 2008-10-08 at 18:51 +0100, Russell Brown wrote:
Can anyone point me at a code fragment (C would be nice) that I could
use to subscribe to hints on a * box?
I'd like to write a small (hopefuly efficient) widget to show custom
device states
Hello everyone,
Since I've been working with SIP more and more I've discovered there
are still plenty of interop and configuration issues between various
pieces of equipment in the real world.
I enjoy helping with SIP issues in this forum and others but I
thought it would make more sense to
Can someone tell me what I am doing wrong? Why doesn't CDR(duration)
or CDR(billsec) return the correct values?
cdr.conf
endbeforehexten=yes
extensions.conf
[macro-Dial]
; ${ARG1} - Dial String
exten = s,1,Dial(${ARG1},,M(post-dial))
exten = h,1,NoOp(Call was hung up - ${CDR(duration)}
A very good idea. I heartily endorse.
Kristian Kielhofner wrote:
Hello everyone,
Since I've been working with SIP more and more I've discovered there
are still plenty of interop and configuration issues between various
pieces of equipment in the real world.
I enjoy helping with SIP
Eric Chamberlain wrote:
Can someone tell me what I am doing wrong? Why doesn't CDR(duration)
or CDR(billsec) return the correct values?
cdr.conf
endbeforehexten=yes
extensions.conf
[macro-Dial]
; ${ARG1} - Dial String
exten = s,1,Dial(${ARG1},,M(post-dial))
exten =
On Fri, Oct 10, 2008 at 08:10:39AM +1930, Luis Morales wrote:
Mike,
Can you tell us :
- asterisk version
- zaptel version
When you call over this line, when you hangup did you hear an busy
tone ? or any class tone ? To do this test connect your lines to
analog phone and make a call.
I've got a situation where I need to use a transfer to the parking lot
as hold, but am not going to use BLF indicators on the phone to pick up
the parked calls so I need to hear the 3-digit extension after the
transfer. I'm using Snom 300 phones and have tried setting a
programmable button to
Hi All,
I'm thinking of creating a new asterisk server using the latest 1.4
stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its
been a while!).
My only concern - my phones are Cisco 7960's (with sccp firmware 7.2
loaded) and to support them better, I remember compiling in a
On Thu, 2008-10-09 at 14:09 -0700, Eric Chamberlain wrote:
Can someone tell me what I am doing wrong? Why doesn't CDR(duration)
or CDR(billsec) return the correct values?
cdr.conf
endbeforehexten=yes
Eric--
To fix a problem, I had to reshuffle things around in the
asterisk core;
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier post
to this list which suggested make uninstall and make remove in the zaptel
directory which just generates errors
On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse [EMAIL PROTECTED] wrote:
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier post
to this list which suggested make
Good point.
I have a T100P that will not be seen by DAHDI for anything, but works
fabulously with Zaptel.
Steve Totaro wrote:
On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The information (or lack of it) on upgrading from zaptel
On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote:
I've got a situation where I need to use a transfer to the parking lot
as hold, but am not going to use BLF indicators on the phone to pick
up
the parked calls so I need to hear the 3-digit extension after the
transfer. I'm using Snom 300
Well,
Your case it's not easy. Try with ringtimeout=8000 (zapata.conf)
option. So it's very strange no signal on hang up. I have an question
for your test. Did you hang up first in your mobile and listen on the
phone for the signal ?
There are other option to play such as:
-
I have four Polycom 330 phones connected to an asterisk system. There
are other VoIP phones connected too. All of the extensions are four
digits beginning with 11. From any of the phones, except the Polycom,
picking up the handset to call extension 1103 for example works fine.
With the
You need to investigate the digit map in the phones configuration. This
determines what dialing patterns the phone will accept. See deails
here:
http://sipx-wiki.calivia.com/index.php/Digit_Maps_used_to_Define_the_Dia
l_Plan
Michael
--Original Message Text---
From: Ed DeHart
Date: Thu, 9 Oct
Ed DeHart schrieb:
I have four Polycom 330 phones connected to an asterisk system. There
are other VoIP phones connected too. All of the extensions are four
digits beginning with 11. From any of the phones, except the Polycom,
picking up the handset to call extension 1103 for example
Ed,
Sounds like the digitmap on the Polycom phones is the issue. You can read more
about the digitmap from:
http://www.voip-info.org/wiki/view/Polycom+Phones
Digitmap reference
Example: [2-9]11|0T|011xxx.T|91[2-9]x|[1-8]xx
It means the following:
* [2-9]11: 911 rule: x11 are
Hi All,
I'm stuck and need some help. I have installed the Asterisk CDR
Analyser Version 2.0.1. It mostly works except for the CDR Report. I
get the following error even though it lists the CDR details.
Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day,
sum(duration)
On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse [EMAIL PROTECTED] wrote:
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier post
to this list which suggested make
On Thu, Oct 9, 2008 at 10:45 AM, Javier Prieto Gomez
[EMAIL PROTECTED] wrote:
Hi I have a Panasonic TDA600 conected with one E1 to the pstn and one PRI
with my Asterisk using a Digium TE220B card.
The Panasonic is master clock and Asterisk is slave, additionally the
Panasonic takes the clock
Remco Barendse wrote:
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier post
to this list which suggested make uninstall and make remove in the zaptel
directory
check digimap into polycom web interface and check the digmap rules
for your voip system
On Fri, Oct 10, 2008 at 9:07 PM, Ed DeHart [EMAIL PROTECTED] wrote:
I have four Polycom 330 phones connected to an asterisk system. There are
other VoIP phones connected too. All of the extensions are
That query appears in call-log.php around line 232.
On Thu, Oct 9, 2008 at 10:20 PM, Klaverstyn, David C
[EMAIL PROTECTED] wrote:
Hi All,
I'm stuck and need some help. I have installed the Asterisk CDR Analyser
Version 2.0.1. It mostly works except for the CDR Report. I get the
Steve Anness wrote:
Thanks for the all the help, I have been pulling my hair out
I now have the trunk working in both directions. However, how do I add
voicemail capability?
exten = _11XXX,1,Dial(iax2/colo/${EXTEN:2},20,Ttr)
exten = _11XXX,n,Voicemail(${EXTEN:2:3}|su)
Thinking that
On Thu, Oct 9, 2008 at 10:32 PM, sean darcy [EMAIL PROTECTED] wrote:
Remco Barendse wrote:
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier
post
to this list
On Thu, Oct 09, 2008 at 08:12:46PM -0400, Alex Balashov wrote:
Good point.
I have a T100P that will not be seen by DAHDI for anything, but works
fabulously with Zaptel.
Which driver does it use?
Is it shown by dahdi_hardware / zaptel_hardware ? (for zaptel_hardware:
try in zaptel 1.4.12
Brilliant, many thanks. It is now working once I changed it to the
correct table name. Line 232 is correct as well, column 103
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Bright
Sent: Friday, 10 October 2008 12:44 PM
To: Asterisk Users Mailing
In download dated 10/9.
Bug fix? Mistake?
sean
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To UNSUBSCRIBE or update options visit:
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You can find the changelog in the the downloads area:
http://downloads.digium.com/pub/telephony/asterisk/
Excerpted from http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.6.0.1
:
2008-10-08 Russell Bryant [EMAIL PROTECTED]
* Asterisk 1.6.0.1 released.
*
I assume you guys are using 1.6.0, yeah? Looks like there was some
sort of confusion about dahdi in 1.6.0... I just saw this because of
Sean Darcy's question about 1.6.0.1 in a different thread. This is
from the 1.6.0.1 changelog:
2008-10-08 Russell Bryant [EMAIL PROTECTED]
*
On Thursday 09 October 2008 09:57:30 pm Steve Totaro wrote:
Now I have not touched any of that code, but to me, it would have been much
simpler to change names, then change functionality later. Make DAHDI a
drop in replacement for Zaptel, in fact, if memory serves me correctly that
is what
Hi all,
I have asterisk 1.4 installed on 2 Dual Core processors (total = 4
cores) + CentOS5.0 x64 + 3 E1 PRI and 80 sip extension. Sometime the
system seems to slow down during peak hours. I check with 'sar' and
see that CPU0 has highest load while the other CPUs has very low
loads.
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