On a production system, running 1.4.17 (compiled from bristuff-0.4.0-test6-xr1)
we had this strange issue two times in the last
weeks:
[2009-01-13 13:58:30] WARNING[1213] channel.c: Fixup failed on channel
SIP/2332-081d0108MASQ, strange things may happen.
[2009-01-13 13:58:30] WARNING[1213]
Perhaps this would help...
http://blog.michaelfmcnamara.com/2007/10/dhcp-options-voip/
Gives details on the dhcp option string needed for the phones and explains
that without it the phone will not accept a DHCP response...
d
2009/1/24 Joseph syscon...@gmail.com
Thanks for the input.
Yes, I
wich limitations?
why you dont just answer the incoming calls in TEST context?
give mucho more info so we can help you.
David
2009/1/24 Pezhman Lali pezhman_l...@yahoo.com
Dear,
because of using dial(local/...) each incoming calls (_12X.) makes 4 ports
on asterisk.
I can not use goto ,
2009/1/22 Laurent Bonny laurent.bo...@gmail.com
Hello,
I am trying to connect an asterisk 1.6 to a trunking plate forme. With
asterisk 1.4.x I added to sip.conf a line asking for registration in the
form of:
register =
and what about add a custome field or setup a variable on outgoing calls and
use the common cdr and then filtering by that field.
David
2009/1/24 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Friday 23 January 2009 18:22:16 Pascal Bruno wrote:
Is it possible to log just the outgoing
Hi,
As you may know, these ISDN BRI features are very important here in Europe
as ISDN Basic Rate Access is very popular among Small Medium Entreprises.
I don't really know why but it seems that in many countries, default is to
install small PBX using Point-to-Multipoint (PtMP) mode as opposed
Copy paste from freeswitch.org
Asterisk uses a modular design where a central core loads shared objects to
extend the functionality with bits of code known as modules. Modules are
used to implement specific protocols such as SIP, add applications such as
custom IVRs and tie in other external
All;
I have a question regarding the Astdb. When reading more than a few values,
it can
take quite a while to grab several
values in the astdb using say, asterisk -rx database show
output.txt and work with that and then set a new value such as asterisk
-rx database put $key $value. The whole
external DB? like mysql?
2009/1/24 cbbs...@hotmail.com
All;
I have a question regarding the Astdb. When reading more than a few
values, it can take quite a while to grab several values in the astdb using
say, asterisk -rx database show output.txt and work with that and then
set a new
Olivier wrote:
Hi,
As you may know, these ISDN BRI features are very important here in
Europe as ISDN Basic Rate Access is very popular among Small Medium
Entreprises.
I don't really know why but it seems that in many countries, default is
to install small PBX using Point-to-Multipoint
Hello everyone!
This is my problem: I try to do gtalk, but my asterisk server uses the local
IP 127.0.0.1 or perhaps the 192.168.*.*.
Now I've heard, that a NAT router can help there. I was told it's the way
the windows-world does the trick, when they sit behind a
router/phonebox/modem.
If you set the 'bindaddr' to your private IP address, the Gtalk
connection from your Asterisk server to my Gtalk client (running on
Windows) works fine. That's at least what we've tested together
Julien, right?
If the STUN packets are properly exchanged between Asterisk and the
Gtalk client
That is a good idea too, where would I configure asterisk to log the channel
status on that custom field?
On Sat, Jan 24, 2009 at 8:27 AM, David fire ddf...@gmail.com wrote:
and what about add a custome field or setup a variable on outgoing calls
and use the common cdr and then filtering
I second that, while read an berkeley db file outside of it's main
application
can work fine, writing in it would certainly lead to huge trouble (data
loss, corrupted file, ...)
A berkeley db file is .. a file, not a database server
David fire a écrit :
external DB? like mysql?
2009/1/24
Hi,
Are you having problems with sip calls or just using Gtalk?
If you are behind a nat router you may need to forward in to your server
port 5252.
Check out the /etc/asterisk/gtalk.conf and /etc/asterisk/jabber.conf files.
Tom
-Original Message-
From:
Does anybody know if idle-url works for Cisco 79xx using Sip? If it doesn't
work is it a Sip vs SCCP issue or Asterisk vs CallManager issue? Thanks
Paul
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi List;
If any one faced the following problem and can help me:
My zaptel version is: 1.4.10.1
My asterisk version is: 1.4.19.1
OS: Fedora core 8
I used make config for the initialization script.
Now, sometimes when the hardware restarted, we discovered that no tone in the
handset
I run chan_sccp at home. It works well, supports the park function, but
does not make use of the conference button. I haven't used the chan_skinny,
so I don't know how it compares. With chan_sccp, if you make a change to
the configuration, you need to reload the module, thus taking down all
Is Zaptel no longer available?
I returned to a long shelved project (using TDM400P and a customized, canned
version of *) and, getting to the configuration, find wctdm is not there. I
recall the authors where very enterprise oriented and focused on T1 cards.
So they left analog support out.
Matthew Fredrickson wrote:
[snip]
I actually was the one that did a lot the work in adding the BRI support
to libpri/chan_dahdi.
[snip]
To answer your final question, for now, if you need NT-PTMP mode, you
should use mISDN.
Hi Matthew,
Is there a BRI status document? I'm asking because
I'm using gtalk.
So I can try to configure my router (it's got a lot of javascript :-) ) to
forward 5222 to my server and the same thing backwards?
Thanks for responding so fast!
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
I'm not completely sure about the things my router can do. It's from the
telephone company and it's supposed to do a lot of stuff. I've just heard,
that windows people could solve such things. After all my setup isn't too
strange or rare? Or is it for running asterisk?
Kndest regards and
The short answer to your question--assuming it is the right question to
be asking--is that Linux comes with a built-in NAT infrastructure as
part of its packet filter (netfilter). The utility iptables is used
to manage it. Simple example:
echo 1 /proc/sys/net/ipv4/ip_forward
iptables
No, your setup is not unusual for a client.
If you are not happy with your router, you can set it to Ethernet bridge
mode (if it's DSL over ATM transport, that's RFC1483). Then your PC
behind it can hold the public Layer 3 interface, but the DSL modem will
still do the ATM/G.DMT stuff.
Patrick wrote:
Matthew Fredrickson wrote:
[snip]
I actually was the one that did a lot the work in adding the BRI support
to libpri/chan_dahdi.
[snip]
To answer your final question, for now, if you need NT-PTMP mode, you
should use mISDN.
Hi Matthew,
Is there a BRI status document?
Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box
that'll work right? The one included by default only deals with debian and
redhat, and the changes between the old zaptel script I have that works are far
too invasive. Notably in the use of this action command that's
On Sat, Jan 24, 2009 at 02:30:24PM -0500, j...@j4computers.com wrote:
Is Zaptel no longer available?
Aparantly no longer linked from asterisk.org . Still very much available
from http://downloads.digium.com/pub/zaptel/ as before.
I returned to a long shelved project (using TDM400P and a
On Sat, Jan 24, 2009 at 11:00:58AM -0500, cbbs...@hotmail.com wrote:
All;
I have a question regarding the Astdb. When reading more than a few
values, it can
take quite a while to grab several
values in the astdb using say, asterisk -rx database show
output.txt and work with that and
On 1/24/2009 at 4:20 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Jan 24, 2009 at 02:30:24PM -0500, j...@j4computers.com wrote:
Is Zaptel no longer available?
Aparantly no longer linked from asterisk.org . Still very much available
from http://downloads.digium.com/pub/zaptel/
While browsing about, found http://www.voip-info.org/wiki/view/TDM400P, where I
found this comment:
Here's a tip passed on from an old telephone engineer. Where your copper
2-wire cable approaches the building, underground, finish with several large
loops, about a metre in diameter, laid on
For fiber installations, be sure that your loops are not placed where
flashes will distract drivers or people performing potentially dangerous
activities.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From:
Hi,
I've it up and running on OpenSuse 11. I used the scripts provided by the
sources and commented out one line:
#
# Determine which kind of configuration we're using
#
#system=redhat # assume redhat
system=debian # assume debian
This forces the script to use debian style. It works for me,
To be fair they did specify underground ;)
j
On Sat, 24 Jan 2009, Don Kelly wrote:
For fiber installations, be sure that your loops are not placed where
flashes will distract drivers or people performing potentially dangerous
activities.
--Don
Don Kelly
PCF Corp
People Come First
Hello:
Yes, DTMF can be a problem on the phones themselves as Sam observed, and inband
can help with this in certain situations. I have DTMF working internally in my
pbx just fine though.
The problem here is transmitting dtmf from my pbx through a carrier to a party
who has phoned into my
Jeff LaCoursiere wrote:
To be fair they did specify underground ;)
j
On Sat, 24 Jan 2009, Don Kelly wrote:
Well sounds like the info was being passed along by someone who did not
understand the purpose.
I would make the loops tighter, and the point is it acts like a choke,
especially
This is, hopefully, just a case of brain fade.
With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and
all LEDS on the card lit, I get no dial tone, plugging an analog phone into
ports 1 or 2, only a buzz and click.
zaptel.conf -
defaultzone=us
loadzone=us
fxoks=1,2
Thanks for the answers. I have to read those more carefully, when I'm properly
awake and concentrated, but it sounds as if this might be of help.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
Stared at an init script long enough, and managed to devise up the following
script. This applies straight to tools/dahdi.init in dadhi-linux-complete.
Minus the top hunk in the patch (which sets system = suse), this converts it
into a working script for suse systems.
Thoughts? What's the
On Sat, 2009-01-24 at 23:45 +0100, Marco wrote:
Hi,
I've it up and running on OpenSuse 11. I used the scripts provided by the
sources and commented out one line:
#
# Determine which kind of configuration we're using
#
#system=redhat # assume redhat
system=debian # assume debian
This
with dahdi I can monitor hardware cards with dahdi show status.
I can then tell if a T1/PRI card goes into condition RED.
When I have a VOIP/SIP connection to lets say Call Manager
how can I monitor this connection?
Today I suddenly started getting 503 service not available messages
when trying
Jerry Geis schrieb:
with dahdi I can monitor hardware cards with dahdi show status.
I can then tell if a T1/PRI card goes into condition RED.
When I have a VOIP/SIP connection to lets say Call Manager
how can I monitor this connection?
Today I suddenly started getting 503 service not
hello! i'm new to asterisk.
i'm using CentOS 5.2 + ASterisk 1.6
when i finish installing asterisk, i configure sip.conf like:
[4455]
type=friend
username=4455
secret=1234
host=dynamic
context=internal
[4466]
type=friend
username=4466
secret=1234
host=dynamic
I am experiencing choppy sound when I bridge from a SIP peer to an IAX
peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am
experiencing choppy sound from the SIP peer to the IAX peer but not
vice-versa. I know that this is not a bandwidth issue because I don't
have choppy sound
Christopher, did you receive the email that I sent to your yesterday?
It was delivered Jan 23 20:47:31 -0600. Maybe it went to your junk box..
I will try again.
Sam
Christopher Gray wrote:
Hello:
Yes, DTMF can be a problem on the phones themselves as Sam observed, and
inband
can help
Muiz Motani wrote:
I am experiencing choppy sound when I bridge from a SIP peer to an IAX
peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am
experiencing choppy sound from the SIP peer to the IAX peer but not
vice-versa. I know that this is not a bandwidth issue because I
Hey,
I am trying to work through a use case requirement where a user
listens to a some advertisement and then if at the end off it they
press a key they press a 1 key they get transfered to a pre-defined
number. I am using the asterisk java library at http://asterisk-java.org/
.
I
Jerry Geis wrote:
with dahdi I can monitor hardware cards with dahdi show status.
I can then tell if a T1/PRI card goes into condition RED.
When I have a VOIP/SIP connection to lets say Call Manager
how can I monitor this connection?
Today I suddenly started getting 503 service not
On Sat, Jan 24, 2009 at 06:38:58PM -0500, j...@j4computers.com wrote:
This is, hopefully, just a case of brain fade.
With zapata.conf and zaptel.conf in place, asterisk loaded, no dial
plan and all LEDS on the card lit, I get no dial tone, plugging an
analog phone into ports 1 or 2, only a
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