Had recent problems with the list, so checking I get list mails now :)
Sorry for the inconvenience,
Johan
___
Johan Sandgren
Svep Design Center AB
Phone +46 46 192 722
Mobile +46 70 173 4152
Box 1233, 221 05 Lund, Sweden
E-mail [EMAIL PROTECTED]
Website
2008/10/29 Alan Lord [EMAIL PROTECTED]
Olivier wrote:
snip /
I'll reply to the correct thread
[featuremap]
blindxfer = ## ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
atxfer = A
Had recent problems with the list, so checking I get list mails now :)
Sorry for the inconvenience,
Johan
___
Johan Sandgren
Svep Design Center AB
Phone +46 46 192 722
Mobile +46 70 173 4152
Box 1233, 221 05 Lund, Sweden
E-mail [EMAIL PROTECTED]
Website
Hello,
I can't say the Works only if some one is offering unmetered only service
or just doing it for fun. If it metered service like calling cards,
termination or metered DID etc, then this can be* really bad*.
If the Service providers use the Sip-B2BUA inside the Sip-Proxy servers.
then, it
Olivier wrote:
snip /
Alan,
Did you get any success with MWI ?
With mine, Asterisk is getting 481 replies whenever Asterisk sends
NOTIFY updates.
Cheers
I don't think so no.
The lamp blinks when I've missed a call but I don't think it correctly
identifies if there are messages in the
- Original Message -
From: Jerry Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 29, 2008 6:55 PM
Subject: [asterisk-users] Current Open Source Billing Package
After spending a couple hours
To follow up --
pbx_lua from trunk works as advertised when backported to 1.6.
pbx_lua from asterisk 1.6 seems hopelessly broken, and I've given up
on trying to persuade it to work.
___
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Hello All,
I have a request from a prospectieve client to deploy a PBX capacity
that can do up to 3000+ lines within a geographic region similar to a
campus. The client wants analog lines for extensions and maybe VoIP
for some backhaul traffic while the other traffic would be carrid via
E1
Hi all,
I'm facing an issue with my asterisk server when an extension (X-Lite
softphone) tries to register on it...A huge amount of packets is exchanged
between endpoint and asterisk server while the X-Lite is online...Even when
I sign out from X-Lite, the asterisk server continues sending packets
On Wed, 29 Oct 2008, Jeff LaCoursiere wrote:
I've been playing with video phones over the past month or 2.
You've got 3 choices: Bottom-end is Xlite, etc. soft-phones.
Desktop videophones - currently Grandtream GXV3000 and ATL4000's.
Top of the range Polycom video conferencing units.
-Original Message-
Hi Folks,
I'm using the rawplayer program to provide my
music-on-hold, and it works very well, with one small
drawback: every time I reset Asterisk, for any reason, the
MoH resets to the beginning of the list. Since MoH isn't used
that often, it basically
Hello,
looks like Asterisk (at least the Version 1.4.21.2 from debian lenny, which
I use) has a serious Problem with pseudo-realtime mode (-p switch) in
conjunction with recent Kernels.
Whenn using -p Asterisk simply hangs forever on startup.
This is almost certainly related to the CFS
On Thu, Oct 30, 2008 at 10:55:42AM +, Sven Geggus wrote:
Hello,
looks like Asterisk (at least the Version 1.4.21.2 from debian lenny, which
I use) has a serious Problem with pseudo-realtime mode (-p switch) in
conjunction with recent Kernels.
Whenn using -p Asterisk simply hangs
These are requests where one endpoint pings the other to check if it
is still alive.
What is the problem?
michel freiha wrote:
Hi all,
I'm facing an issue with my asterisk server when an extension (X-Lite
softphone) tries to register on it...A huge amount of packets is
exchanged between
Dear Alex,
The problem is that the asterisk server is sending these packets
continuously with no stop and with a negligible duration between packets for
the same extension...My Asterisk server read the extensions from the
database and not from extensions.conf...There is a field in the sip buddies
Tzafrir Cohen [EMAIL PROTECTED] wrote:
Not here (Debian Lenny, kernel 2.6.26-1-amd64, official asterisk
packages 1:1.4.21.2~dfsg-2 ).
Can you please test those? (and report a major bug if this is
reproducable with them as well)
I'm not using a Debian Kernel, thus this looks like a problem
By default, the interval at which the qualify pings are sent is, indeed
quite low.
There is no consequence to disabling it except for the obvious
implication that Asterisk then has no way way of knowing if the peer is
dead without first trying to reach it, every time and with every request.
hello !
If I want to use SIP instead of IAX protocol, how can I resolve the NAT
problem with SIP?
Thank you for answering!
Sincerly
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asterisk-users mailing list
To UNSUBSCRIBE or
On Thu, Oct 30, 2008 at 11:36:26AM +, Sven Geggus wrote:
Tzafrir Cohen [EMAIL PROTECTED] wrote:
Not here (Debian Lenny, kernel 2.6.26-1-amd64, official asterisk
packages 1:1.4.21.2~dfsg-2 ).
Can you please test those? (and report a major bug if this is
reproducable with them as
also check jBilling, its a telco oriented billing system, quite nice, there
is also freeside... it might work out for you
On Thu, Oct 30, 2008 at 3:49 PM, Stephen Wingfield [EMAIL PROTECTED] wrote:
- Original Message -
From: Jerry Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing
Any advise on troubleshooting this:
Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF
Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RED alarm is OFF
It
Tzafrir Cohen [EMAIL PROTECTED] wrote:
I'm not using a Debian Kernel, thus this looks like a problem of a
particular Kernel Option.
Do you use debs?
vanilla kernel (currently 2.6.27.4) and debian kernel-packages to build. I
have now Idea, if my machine will work with the official lenny
OK done:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP#Cisco7911Gfirmware841SR1Sconfigurationex
2008/10/29 OCG Technical Support [EMAIL PROTECTED]
Post it on the wiki! I'm sure I'll need it someday
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL
Thanks for the reply!
Generally that's what I do, script local area codes and prefixes so that
dialing 1 is necessary only for long distance calls. The problem here is
that there are over 1500 area codes and prefixes (DC area) that are required
by the carrier to not be dialed with a 1 (ie, local
Ade Vickers wrote:
-Original Message-
Hi Folks,
I'm using the rawplayer program to provide my
music-on-hold, and it works very well, with one small
drawback: every time I reset Asterisk, for any reason, the
MoH resets to the beginning of the list. Since MoH isn't used
that
Jeremy Mann wrote:
Any advise on troubleshooting this:
Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF
Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel:
Alex
I see a fair bit of separate physical networks because of different
management of phones vs IT. In the old businesses Facilities handles
the communications and IT is playing catchup all the time
So in these businesses where the IT side is swapping switches on a
weekly basis it is safer
Cost per analog port can be quite low if done rightl. I would look at
using Xorcom's 32 port asterisk appliances with additional Astribanks
plugged in and a switch or dundi setup.
On Thu, Oct 30, 2008 at 5:02 AM, Dumpolid Exeplish [EMAIL PROTECTED] wrote:
Hello All,
I have a request from a
Dear All,
I have the below settings on my asterisk server and I need to know if there
is a any problem in a setting regarding performance or security..Please
check and let me know:
Global Settings:
SIP Port: 5060
Bindaddress:IP_ADDRESS
2008/10/30 Alan Lord [EMAIL PROTECTED]
Olivier wrote:
snip /
Alan,
Did you get any success with MWI ?
With mine, Asterisk is getting 481 replies whenever Asterisk sends
NOTIFY updates.
Cheers
I don't think so no.
The lamp blinks when I've missed a call but I don't think it
Hi all,
I try to connect two asterisk-server together. There is a server
(obelix) which receives a call. This call should be transfered to
another server.
In my dialplan at obelix I have the following:
exten = 920622201,1,Dial(SIP/outbound:[EMAIL PROTECTED]:${EXTEN})
exten = 920622201,n,Hangup
hello everyone!
I just got the newest asterisk SVN:
trunk# svnversion
152803
and compiled it. then I made some test-calls.
1. Calling my mailbox. It worked, but quality was not good, in comparison to
1.6.0-beta9.
I called via mISDn.
2. Just call myself.
Result: Ringing and asterisk
I tried successfully esnmp ...
2008/10/29 David [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
When I send email from my local asterisk machine, my IP address get's
RBL'd.
I use msmtp;
http://msmtp.sourceforge.net/
Here is my /etc/msmtprc
account default
host mail.bellsouth.net
Hi. In creating a custom extension, and dialing
SIP/222/333#444, seems the party receives only 333
What should I do to send the # symbol? or better, where can I find that
syntax? Googled a lot, nothing.
Thanks!
--
Rodolfo Alcazar
Responsable red y datos
Deutsche Gesellschaft für
Technische
core show application dial (this is the official application doc)
Pay special attention to the D() option.
Rodolfo Alcazar Portillo wrote:
Hi. In creating a custom extension, and dialing
SIP/222/333#444, seems the party receives only 333
What should I do to send the # symbol? or
On many phones # sends the call.
Rodolfo Alcazar Portillo wrote:
Hi. In creating a custom extension, and dialing
SIP/222/333#444, seems the party receives only 333
What should I do to send the # symbol? or better, where can I find that
syntax? Googled a lot, nothing.
Thanks!
--
I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails
from mantis saying things like a note has been added to an issue etc.,
and yet the issue has not been touched in months and the new note it
is referring to is also months old. Consequently, I never received
these e-mails
On 30 Oct 2008, at 15:31, michel freiha wrote:
Do my work for me
There is a lot more to system security and performance than just
Asterisk config. Perhaps you should do some research on this. voip-
info wiki is good.
___
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Daniel Hazelbaker wrote:
I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails
from mantis saying things like a note has been added to an issue etc.,
and yet the issue has not been touched in months and the new note it
is referring to is also months old. Consequently, I
On Thursday 30 October 2008 11:57:29 am Daniel Hazelbaker wrote:
Is it just me or has mantis been holding onto old e-mail and finally
sending it?
i'm getting them too. even the original your license agreement is accepted
email.
--
Anthony - http://messinet.com -
Does anybody know of a mailing list devoted to SIP device or ATA
issues? This is a pretty high traffic list and I'd like to not clutter
any more than I have to. Is there a polycom list for example?
___
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Hi All
I am trying to setup :
PSTN E1 --- Asterisk--Legacy PBX---Legacy Analog extensions.
I've followed steps using :
http://www.voipinfo.org/wiki/view/Asterisk-Panasonic
i get the green light (sync) on both the 2nd span of digium TE420P (that is
cnnected to the legacy pbx pri
Am Donnerstag, den 30.10.2008, 12:17 -0400 schrieb Rodolfo Alcazar
Portillo:
Hi. In creating a custom extension, and dialing
SIP/222/333#444, seems the party receives only 333
Solved, the problem was on my SPA3102, old dialplan:
(**|*x.|x.|**x.)
and now:
(**|*x.|x.|**x.|xxx#x.)
Thanks!
--
I got this working. For what it's worth, here's what the issue.
The channel wasn't getting created under FreePBX via script. Here's what I
needed to do:
1) Run genzaptelconf to generate the zaptel configs
2) find the channel the port(s) is on.
cat /proc/zaptel/*
3)
On Thu, Oct 30, 2008 at 11:03:03AM -0700, hin lee wrote:
I got this working. For what it's worth, here's what the issue.
The channel wasn't getting created under FreePBX via script. Here's what I
needed to do:
1) Run genzaptelconf to generate the zaptel configs
This generates you
1500 prefixes is not a big number. You can use a little script for it (less
than 50 lines).
With a script connecting to a DB server and looking for the prefix, is a
good solution. This way you don't need to force the user to dial the the
leading 1 (or not to do it), you just look on the DB server
Hi,
I would like to get musiconhold from a sound card. This is because I want to
kind of be a DJ and easily change the music playing, etc. However, I
followed the instructions at
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but no
success. i have
[mycustom]
mode=custom
If I were to guess (with no config files it's really just a guess). I
would think your Dial-plan logic isn't using the right 'trunk group' for
calls.
context=from-pstn
group=0
context=from-legacy
group=4
[from-pstn]
On Thu, Oct 30, 2008 at 1:40 PM, Juan Rodríguez [EMAIL PROTECTED] wrote:
With a script connecting to a DB server and looking for the prefix, is a
good solution. This way you don't need to force the user to dial the the
leading 1 (or not to do it), you just look on the DB server and if it does
[sending to -users and -biz in a slightly different format to broaden
participation]
Summary:
Would you help fund different Open-Source Asterisk enhancements,
bugfixes, or documentation if there was a way to collectively
contribute money towards the effort without a profit margin
Subsequent to some previous E-Mails, I've been trying to dig into the ISDN -
BRI situation a bit more. I have determined that I have a HFC card with
Winbond chip, but I'm not sure what combination of drivers is best or usable.
zaphfc is out because it only supports the cologne chip.
misdn is
On Thu, Oct 30, 2008 at 12:44:40PM -0600, Wilton Helm wrote:
Subsequent to some previous E-Mails, I've been trying to dig into the ISDN -
BRI situation a bit more. I have determined that I have a HFC card with
Winbond chip, but I'm not sure what combination of drivers is best or usable.
Tzafrir,
You are correct! I didn't have to commented out the unused FXO ports. So to
revise my earlier email, I have to do the following:
1) Run genzaptelconf
2) Run cat /proc/zaptel/* to find the channel my line is connected to.
3) Add my channel to /etc/asterisk/zapata-channels.conf
ie.
On Thu, Oct 30, 2008 at 12:56:06PM -0700, hin lee wrote:
Tzafrir,
You are correct! I didn't have to commented out the unused FXO ports. So to
revise my earlier email, I have to do the following:
1) Run genzaptelconf
2) Run cat /proc/zaptel/* to find the channel my line is connected
Thanks for the asterisk restart command. That saved me a few minutes during
each test. As for the genzaptelconf command, it creates zaptel.conf and
zapata-auto.conf but not the zaptel-channels.conf. Zaptel-channels.conf is
blank and doesn't work until I manually add a channel to it. Thanks
About this mailing:
You are receiving this e-mail because you subscribed to MSN
Hi list,
I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2.
To cut a long story short, IAX2 is not tx-ing hangup...
Scenario is composed of two asterisk systems A and B.
A receives calls from IAX users X, Y, Z, etc, does some
validation and forwards them to B, also over IAX.
When B
Jonn R Taylor wrote:
I have been able to repeat the results at other locations. The location that
has 26 pages is a linksys PAP2T our accounting person uses remotely to fax
stuff to the office. The ATA is behind a DIL-625 router with QOS on a DSL
line.
I can send faxes from my test sever
Separate cabling is also useful if the phone system is being deployed by
a separate company - it avoids the 'your computer network is generating
rubbish traffic' arguments. (been there before, sadly)
PaulH
Andrew Latham wrote:
Alex
I see a fair bit of separate physical networks because of
I know a business that tried those phones, and removed them.
They found that Polycom phones were 'more' perfect.
PaulH
Bruno Castelo Branco wrote:
hi
O use around 500 atcom530, they are work perfect
www.atcom.com.cn
Gordon Henderson wrote:
On Wed, 29 Oct 2008, Kev Szaszvari wrote:
I have a Dell SC440 with Centos and Asterisk 1.4.21 and a card openvox D110PG,
T1, when a person calling from the PTSN will listen to them but then begins to
distort the voice I heard that name. I probe the card in another computer and
it works perfectly. Anyone has any idea or help. I'm going
Additional question: are there instances when the incoming call waiting in
the queue is dropped when connected to a waiting agent/local extension?
By the way, incoming call channel is: Local/[EMAIL PROTECTED]
created via Originate
On Sun, Oct 26, 2008 at 10:19 PM, Roi Stork [EMAIL PROTECTED]
Here is the QOS script that I use on my bridge.
http://www.taylortelephone.com/asterisk/astshape
I have also had a very high success rate with
Fax--ATA--SIP--Asterisk--SIP--PSTN and the other way. The fax is a Brother
MFC-440CN.
I have posted most of my hylafax iaxmodem configs and other
Hi,
What function or application do I use to get people to type digits into the
phone and store the value into a variable? The application WaitExten is not
what I want that will jump to the new extension.
I want users to enter a number into the phone and store it as a variable so
I can
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Take a look at Read()
Stu
David Klaverstyn wrote:
Hi,
What function or application do I use to get people to type digits into
the phone and store the value into a variable? The application
WaitExten is not what I want that will jump to
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