On 9/6/07, Mani Nair [EMAIL PROTECTED] wrote:
Hello Guys,
I am unable to make calls to outside number from some of my extensions.
Internally I am able to make and receive calls between extensions and also I
am able to receive call from outside number. Any suggestions?
Then in am
Dear Jared;
I would like to ask if there is a method to let the
output of set sip debug ip to be sent for a file?
Regards
Bilal
Hello, I'm working with our SIP provider to nail
down some call
quality issues
we're having, and they've asked me to provide SIP
debug log files
from our
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Dear Jared;
I would like to ask if there is a method to let the
output of set sip debug ip to be sent for a file?
hi
when iam doing this
i see the server is load is very high
how can i send this traffic or mirror traffic to other server
Dear Guillermo;
Is there an english link that help me in configuration
other than:
http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos
Also, what about ASTCC?
Another issue: a2billing support prepaid billing (so
it can be used for calling cards)?
Regards,
On Sat, Sep 08, 2007 at 02:58:40PM -0400, Hariharan Veerappan wrote:
On 9/6/07, Tzafrir Cohen [EMAIL PROTECTED], rcom.com wrote:
udev is not a prerequirement for zaptel. Debian Sarge uses devfs by
default, and Zaptel supports devfs as well.
since the udev not installed in by the sequence,
On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote:
Hi all,
Have just installed v1.4.11 of Asterisk, but I am trying to have it
start at boot but with no luck.
I have used the make config command but it doesn't start. Any help
would be apreciated, many thanks!
use the command
Hi,
On 2007-09-09 at 13:30 Tzafrir Cohen wrote:
use the command update-rc.d
Also, as always in the case of software that has already been packaged,
it may help to look at the existing package.
What parameter should I use with that command?
Many thanks,
Christian
--
Tzafrir
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Tzafrir Cohen wrote:
On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote:
Hi all,
Have just installed v1.4.11 of Asterisk, but I am trying to have it
start at boot but with no luck.
I have used the make config command but it doesn't start.
Ok, the script is attached...
I'll post it on www.generationd.com when I have a chance.
If you have any updates improvements please email them to me! (The
command line parameter handle is pretty stupid - just grew from testing to
production without cleanup).
MD
_
From: Craig
hi:
i create a dundi environment by the caveman can do it dundi
guide. it works fine.but i want to extend the example for my own
need, so i follow the sample dundi.conf config file comes with
asterisk 1.4.11 source.
i try to use precache and failed, and there seems no one know how
to use it
On Sun, 2007-09-09 at 02:44 -0700, bilal ghayyad wrote:
Dear Guillermo;
Is there an english link that help me in configuration
other than:
http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos
Check the www.asterisk2billing.org documentation page.
--- Mark Michelson [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi,
I setup a queue (number 4050) with one static
agent
(extension 4054).
What I would like is that when someone calls the
4050
queue and there are neither dynamic agents
logged in
nor is the static agent 4054 on-line
Hi,
as noone out there seems to be able to maintain chan_sccp, i'm trying to
switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly
wrong/non functional. I see
Redial NewCall CFwdAll more
(more)
CFwdBu... GPickUp Confrn more
NewCall works, CFwdAll seems to toggle DnD, the
On Sun, 2007-09-09 at 15:45 +, Andreas Anderson wrote:
Hi,
as noone out there seems to be able to maintain chan_sccp, i'm trying to
switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly
wrong/non functional. I see
Redial NewCall CFwdAll more
(more)
CFwdBu...
I applied the patch, however, I'd like to know which particular files to
copy after running a make. I do not wish to run make install as it will
overwrite other configuration changes I have made.
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote:
I applied the patch, however, I'd like to know which particular files to
copy after running a make. I do not wish to run make install as it will
overwrite other configuration changes I have made.
A make install will not overwrite any
Jeremy P wrote:
Thanks for all the good info. If you're looking for a cheaper version
of the thin client you could try the t5530. It's about $300 US but it
only has 64 MB Flash. A 1GB flash module is $70 US but sounds like
overkill for your application.
Frankly, the 70 clams is the
G B wrote:
Hi,
I appreciate the help. I called the vendor of the card and they
recommended removing all of the PCI cards on the system (including the
video card), and moving the card to a new PCI slot.
I did all of them together, ran the system headless, and ssh'ed in
remotely. It
Steve Totaro wrote:
I am the last one to pickup a manual or call tech support but yesterday
I was working on a very large industrial ShopBot (It is a robot so that
is cool and it does really awesome things but why I was working on it
don't ask.. http://www.shopbottools.com/applications.htm
David Gomillion wrote:
On 9/7/07, *Michelle Dupuis* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
This is going into an emergency response facility...where they currently
have a Nortel Option 61 (I think). They want to slowly phase into VoIP.
They will need 1000 phone
I have 4 TDM T1's going in to a IVR system. The IVR messages are
recorded .wav format - The system appears to crap out at about 40 calls
- Would using GSM or some other format help save CPU cycles?
Using 1.2, Dual Xeon and 2GB ram
TIA
--
Barton Fisher
Innovative Communications
714-228-5400
'show channels' shows only running calls while 'sip show channels' shows
all running sip sessions including phones trying to register .
On 09/09/2007, ram [EMAIL PROTECTED] wrote:
Hi all
what is the difference between
show channels
sip show channles
i see the difference in both
show
On 10:28, Sun 09 Sep 07, Barton Fisher wrote:
I have 4 TDM T1's going in to a IVR system. The IVR messages are
recorded .wav format - The system appears to crap out at about 40 calls
- Would using GSM or some other format help save CPU cycles?
Using 1.2, Dual Xeon and 2GB ram
depends on
Hi,
my ATA has two phones attached and the possibility to set different
accounts.
I put two account of my asterisk server, however, it is able to call only
with the second one in order to the sip.conf and the first it gives me 403.
And idea how to solve it?
On 9/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
'show channels' shows only running calls while 'sip show channels' shows
all running sip sessions including phones trying to register .
thanks
but after my 30 channels of show channels
i see lot of vice break and choppy voice
doing
Michiel van Baak wrote:
On 10:28, Sun 09 Sep 07, Barton Fisher wrote:
I have 4 TDM T1's going in to a IVR system. The IVR messages are
recorded .wav format - The system appears to crap out at about 40 calls
- Would using GSM or some other format help save CPU cycles?
Using 1.2, Dual
I was wondering if this bug: http://bugs.digium.com/view.php?id=10535
would affect a PRI connection.
I seem to be dropping DTMF digits on the PRI.
The company says they have test the line and they way the PRI is fine
as far as they are concerned.
So will this bug and patch help me? I am running
Hi,
What parameter should I use to that command?
On 2007-09-09 at 13:45 Ron Wellsted wrote:
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Tzafrir Cohen wrote:
On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote:
Hi all,
Have just installed v1.4.11 of Asterisk, but I am trying to have
/asterisk-users
__ NOD32 2516 (20070909) Information __
This message was checked by NOD32 antivirus system.
http://www.eset.com
--
Barton Fisher
Innovative Communications
714-228-5400 Ext 5410
http://www.icpage.com
begin:vcard
fn:Barton Fisher
n:Fisher;Barton
Dear Ram;
You are able to send it for a file?
Regards
Bilal
Dear Jared;
I would like to ask if there is a method to let the
output of set sip debug ip to be sent for a file?
hi
when iam doing this
i see the server is load is very high
how can i send this traffic or mirror traffic to
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Christian wrote:
Hi,
What parameter should I use to that command?
On 2007-09-09 at 13:45 Ron Wellsted wrote:
Tzafrir Cohen wrote:
On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote:
Hi all,
Have just installed v1.4.11 of Asterisk,
You can try MOR: www.kolmisoft.com/mor
It does what you need. It does it even in FREE version.
PRO version costs _many_ times less then other not free solutions mentioned
in this thread.
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
From: [EMAIL PROTECTED]
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
Hi,
my ATA has two phones attached and the possibility to set different
accounts.
I put two account of my asterisk server, however, it is able to call
only with the second one in order to the sip.conf and the first it
gives me 403.
Hello,
On 2007-09-09 at 22:36 Ron Wellsted wrote:
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Christian wrote:
Hi,
What parameter should I use to that command?
On 2007-09-09 at 13:45 Ron Wellsted wrote:
Tzafrir Cohen wrote:
On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian
I have searched this list and others, and see other pepole having this
issue. However, I have not seen how to fix it.
Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
retries exceeded on transmission
778f89593967725f0abe40eb1752504c for seqno 1620 (Critical
Hi List;
What is the difference between increasing the verbose
level and the debug level?
By increasing the verbose level, then I will get more
traces messages and by increasing the debug level, I
will also get more traces messages. So what is the
difference?
Any help?
Regards
Bilal Ghayad
Hi List;
If I set nat=yes, then asterisk will send the packets
to the public IP address or to the private IP address
(which will be for the endpoint that is behind the
nating)?
And by setting the nat=yes, then what exactly will be
ignored at asterisk side when reading the
registeration messages
Hi List;
If I need traffic to be directly between the
endpoints, then I have to set the canreinvite = yes?
If I did not configure the canrenvite at all, then by
default it will pass the traffic via Asterisk and not
directly between the endpoints?
What if one endpoint was SIP and configured with
Hi List;
Did any one see a SIP endpoint that can work without
need to do registeration on the gatekeeper side? If
this SIP endpoint existed, then I can configure the
host=static, but I am not able to find any SIP
endpoint accept to not register (all sip endpoints
request to register), but most of
Mediatrix 1204, and i assume their other models as well
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
Did any one see a SIP endpoint that can work without
need to do registeration on the gatekeeper side? If
this SIP endpoint existed, then I can configure the
host=static, but I
By default assuming you have no global setting otherwise, if asterisk
doesnt see a need to stay in the path then it wont. hence if it has to
transcode between different codecs, capture DTMF or different
protocols it will stay in the path.
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi
If you set yes then asterisk assumes that the address its coming from
is not the same as the UA thinks it is. most devices will not operate
properly if set to yes when they are in fact local.
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
If I set nat=yes, then asterisk will send
Check it out in the CLI and you will see for yourself.
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
What is the difference between increasing the verbose
level and the debug level?
By increasing the verbose level, then I will get more
traces messages and by increasing the
In general keep in mind, asterisk is very user friendly and wont bite
:). Trial and error is a good friend to get to know asterisk so that
you know what all of these mean.
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
What is the difference between increasing the verbose
level
BTW, AFAIK, there is no such thing as host=static it's either dynamic
or an IP/Name.
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
If I set nat=yes, then asterisk will send the packets
to the public IP address or to the private IP address
(which will be for the endpoint that is
I suspect if you remove the callerid entry from this device's
sip.confdefinition things will work better.
On 9/9/07, Apa Minerala [EMAIL PROTECTED] wrote:
I have searched this list and others, and see other pepole having this
issue. However, I have not seen how to fix it.
Sep 6 18:52:36
Thanks. My results after applying the patch and recompiling are that the
problem can only be replicated with calls from mobile networks. Digits like
160 entered in the digital receptionist by a caller are received by the
asterisk server as 16660 sometimes. Other times it is received as 1660.
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
Well, it seems there are differences between those accounts then.
You might want to post your sip.conf, and -if that is possible- the ATA
conf file; or at least a
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Justin Ridge wrote:
Hi all,
Configuration: Analog phone connected to TDM400p.
I'd like the phone to give a half-ring (chirp) periodically when there
is a message waiting. Can this be done? How is it configured?
The visible Message
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Rizwan Hisham wrote:
hi all,
Is there any default timeout for manager connection. If its configurable
then plz tell me how.
In the sample manager.conf file there is the following:
; If the device connected via this user accepts input slowly,
;
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Barton Fisher wrote:
Thanks, OK, a bit confused The cards are TE410P. I really don't
see how the set a codec for this, other than it might default to
something in code like ulaw. Any clue on how to verify codec in use
during a call?
C F, I have nat=yes set by default for all my extensions(with
canreinvite=no). And things work fine.
Bilal, about Asterisk sending packets to public/private :
Asterisk will send packets to the public IP advertised by the msg/recv
from address. It is the NAT's headache on the endpoints network
Sir,
I am having Asterisk pbx which is running without any problem now i want to
connect this with Panasonic pbx with FXS port so, if any body want to call
panasonic users than he will call or vise-versa. i want to connect only two
extension with Asterisk so, all communication done only on
Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED]
wrote:
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
Well, it seems there are differences between those accounts
then.
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