On Thu, Aug 31, 2006 at 06:10:30PM +0200, Giorgio Incantalupo wrote:
Hi Julian,
I'm using beronet BRI cards which are good and have autoconfiguring sw
for installation. (I tried junghanns bristuff and I had more problems to
install but maybe it is been improved lately). The only little
As do mine. I wasn't saying I was missing files, I was saying that
the file which was being requested by the phone actually exists on
the TFTP server (it was in the SIP .cop file), but the phone doesn't
pick up the file it's asking for, even though it exists.
On 1-Sep-06, at 1:38 PM, Lacy
With respect, the problem is with your numbering plan..
CP
On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:
I found a problem in blind transfer:
I have an extension number 601 and I have an extension 6014
If I get a call on 615 (snom) and transfer to 6014 it works, since
snom
Hi all,I am using MySQL for mysql_cdr. I have very strange issue, while destination is ringing and caller disconnect the phone without any conversation, i can see in cdr of mysql the duration is starting and for this customer are charged without any calls.Any can suggest me how i can stop this
Anthony Rodgers wrote:
With respect, the problem is with your numbering plan..
WHERE do you see a problem in the numbering plan?
I see the problem in ASTERISK, because it does not wait for the last
digit!!!
Where can I set that it waits for it?
The beauty on voip IS that you can
Hi,
I heard of a news, that there is a replacement codec available for
g729 and accept the g729 codec data for decoding. Anyone familier with
this? Also the good news is that it is noted that it works fine with
asterisk and the g729 encoded data.
Anyone has the link for the free asterisk
Hello all,
For some reason when dialing in I get the IVR or if I forward
to my conference line... any keys pressed seem like they arent received
.. Like Im pressing them, but they arent being registered with
the server .. Any ideas?
Im using the vmware nerdvittles build, the latest
You need to install libmysqlclient15dev, it's saying it can't find the
header files it requires.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher Aloi
Sent: Friday, August 25, 2006 8:36 PM
To: Asterisk Users Mailing List - Non-Commercial
Try changing the DTMF mode for that line, I've found that if rfc2833 doesn't
work, inband will
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lenny
Sent: Saturday, September 02, 2006 4:28 AM
To: Asterisk-Users@lists.digium.com
Subject:
Hello,
So make the changes to what part in FreePBX?
Thanks..
LB
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rushowr
Sent: Saturday, September 02, 2006 5:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Abdul wrote:
Hi all,
I am using MySQL for mysql_cdr. I have very strange issue, while destination
is ringing and caller disconnect the phone without any conversation, i can
see in cdr of mysql the duration is starting and for this customer are
Note that the other end also has to make the change as well, so you
need to talk to them unless its yours.
on Saturday 09/02/2006 Lenny([EMAIL PROTECTED]) wrote
Hello,
So make the changes to what part in FreePBX?
Thanks..
LB
-Original Message-
From: [EMAIL
On Sat, Sep 02, 2006 at 12:50:27AM -0400, Dean Collins wrote:
Lol Tzafir, you posted more than a few xorcom posts here (which I was
very appreciative as I used to use it before [EMAIL PROTECTED] was built).
Xorcom Rapid is Free software[*]. Unlike other products, it is not a direct
source of
Nope.
It just uses intel libraries.
I'm taking about a codec not g729. But can accept g729 encoded data
and also product clone of g729 encoded data.
It is a replacement of existing g729 where it uses a different
algorithm different from the original one.
Assume you have unlimited channels
Nothing new I can see.
We have the same problem with the address book, we are preferring to
use the XML services to present an address book rather than the built
in one.
On 31/08/2006, at 7:47 PM, Tomislav Parčina wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Seems
On Fri, Aug 25, 2006 at 08:35:59PM -0400, Christopher Aloi wrote:
Hello All -
Running the following:
Debian Stable
Asterisk SVN-branch-1.2-r41069
Checked out the following from SVN:
asterisk-addons/branches/1.2
When I attempt to compile asterisk-addons I get the following:
The
Nope.
It just uses intel libraries.
I'm taking about a codec not g729. But can accept g729 encoded data
and also product clone of g729 encoded data.
It is a replacement of existing g729 where it uses a different
algorithm different from the original one.
Assume you have unlimited channels
On Thu, Aug 31, 2006 at 09:40:50PM -0600, Michael Welter wrote:
My Asterisk colo server has been up for almost two years. Today it
crashed. When I gave the reboot command, it crashed so hard that it had
to be power cycled. I wasn't in attendance, but I can speculate that it
had a kernel
Ronald,
You seem to be a little bit angry about VoIP. If so, I could give
you my old Nortel system. Does this would make you happy?
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 2 septembre 2006 04:20
À :
On Thu, Aug 31, 2006 at 05:58:49PM +0200, Noc Phibee wrote:
Hi
when i want compile asterisk 1.2.11, i have this error :
make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime'
cd editline unset CFLAGS LIBS test -f config.h || CFLAGS=-O6
./configure
loading cache ./config.cache
Do not know of a card that does. But think a digium T1 to a channel
bank (ie Adit600) would.
On Sep 1, 2006, at 2:06 PM, Tim Sharp wrote:
I am looking at CTPX's VP2000 product. I haven't tried it yet.
Please let me know if you find a solution that works.
Tim
-Original Message-
Elpidio,
Glad to hear it. Depending on your config, you may need to allow the
RTP ports through as well. I poked holes in my firewall for ports
1-2, probably overkill, though.
Bob...
Elpidio Ramos wrote:
This helped a lot.
It was the firewall. I got it configured right now.
On 1 Sep 2006, at 16:11, Jay Milk wrote:
Asterisk is the least of your problems here. You first need to
talk to your country's telephone operator and ask if it's possible
to get a toll-free prefix or area code. IF they can, I'm sure
they'll charge you handsomely for that privilege --
Hello,
This is speaking about outside callers. How can I tell them to change their
DTMF; I'm sure I can't.
Where can I though make the changes initially in PBX/trixbox?
Thanks..
LB
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent:
I'm thinking this might be my problem ..
http://forum.stanaphone.com/viewtopic.php?t=3149highlight=callers+press+key
s
Poster suggests:
What service are you using to call your StanaPhone number? If it is another
VoIP service it is possible that your DTMF keys are being swallowed by that
leg of
Has anyone seen or played with the Nokia N80 yet? http://news.yahoo.com/s/pcworld/20060831/tc_pcworld/126998
I noticed people saying they were having difficulties
connecting asterisk and the N60/N61 and wondering if this will have solved the
problems?
Im about to pull the trigger on
Hi Corey,
I spend 2 hours with REALLY bad docs on how to Unlock the PAP2 Vonage I
got from Office Max. I bought it the second I saw a glimpse of an articaly
that it could be turned back into an NA. Anyone want to try this? The nes
ones one the shelf in my area had 3.1.3 code already, but if
CFLAGS ... looks like Gentoo ..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Saturday, September 02, 2006 8:54 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problems compil 1.2.11
On Thu, Aug 31, 2006 at
Ciao Noc,
checking whether the C compiler (gcc -O6 ) works... no
In my gcc version (3.4.4), there's no -O6 switch.
Try removing from your CFLAGS the -O6 switch, or replacing it with a
more conservative -O2.
HTH,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
David Gagnon wrote:
Ronald,
You seem to be a little bit angry about VoIP. If so, I could give
you my old Nortel system. Does this would make you happy?
David
David,
I am not angry about VoIP, but please send my your old Nortel system !
I just do not understand why I can
Lenny wrote:
Hello all,
For some reason when dialing in I get the IVR or if I forward to my
conference line... any keys pressed seem like they aren’t received ..
Like I’m pressing them, but they aren’t being registered with the
server .. Any ideas?
I’m using the vmware nerdvittles build,
Hello Ronald ..
This is what I'm trying to learn of now ..
Where in freepbx do I place these settings?
Trunk settings?
If I could just get that bit of info..
Thanks
LB
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Saturday,
Lenny wrote:
Hello Ronald ..
This is what I'm trying to learn of now ..
Where in freepbx do I place these settings?
sip.conf ;-)
that was easy, ... do you have another question?
bye
Ronald
Trunk settings?
If I could just get that bit of info..
Thanks
LB
-Original Message-
In freepbx, its in the peer details of the trunk.
on Sunday 09/03/2006 Ronald Wiplinger([EMAIL PROTECTED]) wrote
Lenny wrote:
Hello Ronald ..
This is what I'm trying to learn of now ..
Where in freepbx do I place these settings?
sip.conf ;-)
that was easy, ... do you
Yes, I have tried a few ways in the sip.conf
I have tried inband and rfc2833 and auto ..
As I said before...
++
I'm thinking this might be my problem ..
http://forum.stanaphone.com/viewtopic.php?t=3149highlight=callers+press+key
s
Poster suggests:
What service are you using
Nick,
I've used a SPA3000. There seems to be a later model from Linksys,
hopefully it works better. I had some severe echo problems due to my
distance from the CO. The SPA3000 never could seem to compensate. The
older software worked better, but it never passed muster with the wife.
Went to
** See my last email..
Ronald suggested in the sip.conf
You suggest the peer details .. That would be for the outgoing settings;
isn't this a incoming handler?
Anywho .. none of the suggestions worked..
Check my last email as a potential culprit might be the connection im
using..
What are
PLEASE DON'T CROSS POST!
Kannaiyan Natesan wrote:
I heard of a news, that there is a replacement codec available for
g729 and accept the g729 codec data for decoding. [...] If there is
any royalty need to pay, is that cheaper than the existing g729
cost?.
G729 is not royalty free.
I have never tried this, but what about an analog FXS card, set to use featd
or em_wink signalling? The FXS card will supply battery (digium hardware
actually supplies the appropriate voltages). You would just have to use the
appropriate signalling type to provide the winks.
-Tim
On
John,
Ok .. I'm really under the idea that this is an ISP issue and a conflict
with trying to run VoIP on an already VoIP enabled line..
Thanks for the suggestions...
LB
-Original Message-
From: John covici [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 02, 2006 1:13 PM
To: Lenny
STOP !!
I'm least bothered whether g729 works or not or what the developer did
to made it to work.
I'm bothered how it works and what are the details about it. As you
are keen, I'm also keen on the developer who made that software and
what they claim. Most importantly I don't
Hi all,Im quite new to SPA3000. I have a TRIXBOX running on public address. I need my SPA3000's FXO to be used as a trunk from a dynamic address behind NAT. Is this scenario possible?Please give me some good links if it works.. I would really appreciate any help as my TRIXBOX is in US and my
Please note that this code is available for you to download for
education purposes only and if a patent exists in your country for
G.729 or G.723.1 then you should contact the owner of that patent and
request their permission before executing the code.
It is noted on the website with the above.
Please note that this code is available for you to download for
education purposes only and if a patent exists in your country for
G.729 or G.723.1 then you should contact the owner of that patent and
request their permission before executing the code.
It is noted on the website with the above.
That is very creative way to get to where I want to be ... I would not have
thought of that angle ... I will give this a try ... Thanks ...
G.Hendershot
-Original Message-
From: Tim St. Pierre [mailto:[EMAIL PROTECTED]
Sent: Friday, September 01, 2006 10:43 PM
To: [EMAIL PROTECTED];
On Sat, Sep 02, 2006 at 05:50:09PM +0200, Andrea Spadaccini wrote:
Ciao Noc,
checking whether the C compiler (gcc -O6 ) works... no
In my gcc version (3.4.4), there's no -O6 switch.
in the man page of my gcc there isn't either. It will build Asterisk
just the same. I'm not sure if it
Dialing a number and transferring a number are two different things. And
no offense, you are not really providing a lot of details along with
your problem. So you can dial the numbers but not transfer from one to
the other.
What does the CLI say when you try the transfer? That would provide a
Hi Tzafrir,
Actually, it would appear as something is wrong with the PHP script
Michael is referring to. As far as I understand AGI, for each AGI script
that has to be run, asterisk will fork it self out, run the AGI within
the fork, then return back to asterisk once the AGI is complete.
Mulitech make one but 2 port gateway is $1000.00. ouch.
still looking.
Jonn
On Sat, 2 Sep 2006 08:45:47 -0500
Jerry Jones [EMAIL PROTECTED] wrote:
Do not know of a card that does. But think a digium T1
to a channel bank (ie Adit600) would.
On Sep 1, 2006, at 2:06 PM, Tim Sharp wrote:
I
Hey Bob,
I think the SPA31-2 is the new guy on the block. Only $10 more too mail
order. $86 was the best I saw.
So I have the PAP2 with two cheapy $4 wall phones mounted in the kids
room, they are calling each other and my laptop.. Only issue so far is
that to call one PAP2 from the other
on start asterisk -vvvc fails with the following.
[chan_sip.so]Sep 2 13:06:44 WARNING[20066]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined
symbol: ast_osp_terminate
Sep 2 13:06:44 WARNING[20066]: loader.c:554 load_modules: Loading
module chan_sip.so failed!
I have just noticed that my X-Lite soft phones don't dial 3-4 digit
extensions without first dialing it in the display and then hitting send.
So tthat is an issue with the phone you think? Ok, I'll start there for
the inter digit timeout, see if there is a certain dial string lenth
before
About 70% of the time, my Local DID provider sends me ANI II digits
(see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html)
where there will be an extra 2 digits
added to the Caller ID - For example 62714222 where '62' = Cell
Phone for example..
The problem is, I have not
Hi,
While dialing calls formmy client's office, where I was working, the caller ID goes as the extension number of the phone from where caller is calling. I tried to playaround with config files, also changed info in their Grandstream GX-2000 phones, but to no avail. What am I missing here and
Hi everybody,
I'm trying to load-test my Asterisk PBX using SIPP, but I always
getting errors, I followed the instructions given in [1] which mainly
was to create the user sipp in sip.conf and the dialing plan for his
context in extensions.conf
I'm using Asterisk 1.0.10
Any ideas or tutorial on
Hi everybody,
My client had just installed a PRI in his office for his phone line, with 30 DIDs. Main phone number ends in 1900 and DIDs last 4 digits are from 3570 to 3599. Now when caller calls number ending in 1900, call comes in with DID 1900, and asterisk answers it. Second caller calls,
Hi,
My client has all Grandstream GX-2000 phones in his office and he wants receptionist to use them for paging as well. Currently they are using Nortel and receptionist can easilydo paging. He said that he had somebody setup their old Asterisk system in a way, that receptionist could dial an
Hi Folks,
I'm trying to use the Queue feature to essentially implement a
multiple call appearance situation for some of our executives.
Essentially I have a queue defined per executive like:
exten=9495551212,1, Queue(stever|tTr|||25)
exten=9495551212,2, Goto(druid-users,1212,1)
So the user
Nick,
I know some adults that can have an entire conversation in the same
amount of time.
Does pressing the # key speed up dialing? If so look for a timer in the
PAP config or tell the kids to press #. IIRC the spa3k had something
similar, but never did much in-house dialing.
$86 is a
Kevin Smith wrote:
Dialing a number and transferring a number are two different things.
And no offense, you are not really providing a lot of details along
with your problem. So you can dial the numbers but not transfer from
one to the other.
I was not thinking that it would be too much
Zeeshan Zakaria wrote:
My client has all Grandstream GX-2000 phones in his office and he
wants receptionist to use them for paging as well. Currently they are
using Nortel and receptionist can easily do paging. He said that he
had somebody setup their old Asterisk system in a way, that
-Ursprüngliche Nachricht-
Von: Mr. Jones [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 3. September 2006 01:12
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Queue timeout problems
Hi Folks,
I'm trying to use the Queue feature to essentially implement a
multiple
Are you using # to transfer? If so, it's not sending it as a new call, it's
just sending asterisk digits using whatever DTMF mode. Asterisk parses these
based on a first match in the dialplan. Make sure that the longer
extension numbers are loaded first in the dialplan.
-Tim
On September
Somewhere in your outbound routing section of the dialplan, you need to have
this line:
,n,Set(CALLERID(number)=whateverthenumbershouldbe)
Personally, I like to set a variable in sip.conf, perhaps PSTNCALLERID, that I
use in the above line. That way I can set PSTN caller ID numbers on a per
Hey Bob,
Just tested the PAP2, yes a # sends right away.
I am looking for why, still new at the dial plan stuff.. this is the
default.. Should I be looking for a way to have the PAP2 NOT deal with
dialing and let Asterisk handle it?
You probably have to set all your PRI channels as part of a trunk group.
Additional calls to the same number should show the same number. Make sure
that when they hit your dialplan, there is somwhere for a second call to go
(ie. a queue, voicemail, another extension, etc.)
-Tim
On September
At 03:37 PM 9/2/2006, you wrote:
What can I do to strip these digits from Caller ID before answering
the call so CDR and Voice Mail Caller ID announcement show correct number?
I probably mis-typed something here, but something like this should do:
exten =
There is a dialplan setting in the advanced config. If you modify this to
recognize your three or four digit extension pattern, it will dial instantly
after you dial an extension.
-Tim
On September 2, 2006 19:14, Bob Chiodini wrote:
Nick,
I know some adults that can have an entire
You could create a function that uses GotoIf() to detect the extra digits.
The line it points to could strip the extra digits.
What version of asterisk are you using? (the functions are different
pre-1.2.1)
On September 2, 2006 18:37, Bart Fisher wrote:
About 70% of the time, my Local DID
Ok, I found the Interdigit short timer (3 secs) and Interdigit long timer
(sure enough, 10 secs) So, what I have seen is that when a dial plan hits
a match, it fires without looking for more digits.. The interdigit short
delay is in effect, but the long timer hits ya when you are trying to
You have to set in in the PAP2. When using SIP, it has to send an invite with
the number it wants to be connected to. The Sipura has to know a complete
number to send - it can't send it in pieces. You need to make the dialplan
in the Sipura match what you have programmed in Asterisk.
Ie. My
Zeeshan Zakaria wrote:
Hi everybody,
My client had just installed a PRI in his office for his phone line,
with 30 DIDs. Main phone number ends in 1900 and DIDs last 4 digits
are from 3570 to 3599. Now when caller calls number ending in 1900,
call comes in with DID 1900, and asterisk
Nic Bellamy wrote:
Zeeshan Zakaria wrote:
My client has all Grandstream GX-2000 phones in his office and he
wants receptionist to use them for paging as well. Currently they are
using Nortel and receptionist can easily do paging. He said that he
had somebody setup their old Asterisk system
Tim St. Pierre wrote:
Are you using # to transfer? If so, it's not sending it as a new call, it's
just sending asterisk digits using whatever DTMF mode. Asterisk parses these
based on a first match in the dialplan. Make sure that the longer
extension numbers are loaded first in the
Can the FXS card do reverse battery?
Jonn
On Sat, 2 Sep 2006 12:54:34 -0400
Tim St. Pierre [EMAIL PROTECTED] wrote:
I have never tried this, but what about an analog FXS
card, set to use featd
or em_wink signalling? The FXS card will supply battery
(digium hardware
actually supplies the
Hi Tim,
The dial plan trick worked great. Added |40[01]x| to my plan and 4000-4019
connect instantly from the PAP2 :) Added it to my X-Lite as well, and
worked there too.
Thanks!
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
I'm looking for a way to dial my contacts using a SIP or VOIP gateway in Thunderbirds Addressbook. I can do this using Outlook with SIPTAPI, ASTAPI, and a couple of others, however, I have not found a way to do so in Thunderbird.
Anybody have any ideas?
Thank you in advance.
On Sat, 2 Sep 2006, Diego Quintana Cruz wrote:
Hi everybody,
I'm trying to load-test my Asterisk PBX using SIPP, but I always
getting errors, I followed the instructions given in [1] which mainly
was to create the user sipp in sip.conf and the dialing plan for his
context in extensions.conf
Thanks Guido -
I tried that and still have the same problem. The call never seems to
leave the queue.
Any other ideas?
On 9/2/06, Guido Hecken [EMAIL PROTECTED] wrote:
-Ursprüngliche Nachricht-
Von: Mr. Jones [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 3. September 2006 01:12
An:
When ever we do a roll out of Asterisk in a small business environment
replacing an old key system or legacy PBX the receptionist always asks
us, How do I know if someone is on a call before transferring them?.
My typical answer is why do you need to know, just do an attended
transfer and if they
No one has any ideas?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of George A.
Roberts IV
Sent: Friday, September 01, 2006 10:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help with blind transfer
Hello all,
81 matches
Mail list logo
