On 13:16, Fri 22 Dec 06, Rajeev Natarajan wrote:
> I think the + convention started off because different countries have
> different international access codes. Well, on GSM networks, + can be a
> part of the number to represent the international access code ( the
> traditional access code in Indi
Have a look at www.spidermux.org
Zoa
Allen Casteran wrote:
We have an application for Asterisk that will require connecting 144
fax ports into the system. Faxes will route externally over a PRI. The
144 ports are for local fax machines within the building. Not all will
be faxing simultaneous
In article <[EMAIL PROTECTED]>,
Robert Jenkins <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm using Asterisk (1.2.13) on Centos 4.4 x86_64 with a TDM2400E for analog
> trunks (& extensions) plus some Polycom 501 & 601 phones.
>
> I have a problem in that the audio via the Polycoms is gated or muted dur
Conrad Wood wrote:
On Thu, 2006-12-21 at 12:07 -0700, Douglas Garstang wrote:
I'm no C programmer, but is this 32 limit just an array definition somewhere?
Wouldn't it be a no brainer to track it down and increase it so some very large
number?
I think pickupgroup is defined as 'uns
this limitation still persist in 1.4 or trunk? if yes, it should be
really improved...
imagine, even small companies have more than 32 offices, i.e.
callgroups, where phone call pickups are needed.
John Harragin wrote:
callgroups & pickupgroups greater than 31 are not working for sip calls
w
Hi Carlos,
please delete the line "dchan=16".
dchan is use for ccs framing.
You can use the follow:
#
span=1,0,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
#
i hope this help you ;p
--
Humberto Figuera - Using Linux 2.6.18
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG K
Am Freitag, den 22.12.2006, 00:53 -0500 schrieb Doug Crompton:
> Question... What is the purpose of the + before the number? Does anyone
> actually have to enter it? If so how would you do it? It is not used in
> the US but do I see it come in on SIP lines CID. I assume the CID ignores
> it in the
I've a problem to communicate from a trunk to a another trunk.
My config is :
PBX 1 with subscribers 100 - 119 (SIP trunk to asterisk)
PBX 2 with subscribers 200 - 219 (SIP trunk to asterisk)
Asterisk with Softphone 300 - 319
I can call from Softphone to PBX 1 and from Softphone to PBX 2
I can c
Hi,
I created a script named example2.sh which goal is read some text from my HP
Service Desk using an application in java and send this text to the text2wave
application for TTS.
example2.sh
java -Xbatch Example10 | text2wave -f 8000 -o /var/lib/asterisk/sounds/my-sd.wav
When I execute the s
Does the user who is running asterisk has permissions to execute it? check
you script file permissions.
On 12/22/06, Andre Gustavo Lomonaco <[EMAIL PROTECTED]> wrote:
Hi,
I created a script named example2.sh which goal is read some text from my
HP Service Desk using an application in java and
Hi. I am having a strange problem when using the 1.4 version of
asterisk and zaptel. If I call from a pstn line into the asterisk box
using a phone number which calls the box via sip, then once I am in
the meetme conference nothing happens when I hit the star key -- I
cannot get the user menu. T
Post this at bugs.digium.com along with some more info like if it crashes at
use of some specific application or randomly .
On 22/12/06, Edwin Lam <[EMAIL PROTECTED]> wrote:
our * crashed twice in a month with segmentation fault &
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt
On Dec 22, 2006, at 3:56 AM, Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Robert Jenkins <[EMAIL PROTECTED]> wrote:
Hi,
I'm using Asterisk (1.2.13) on Centos 4.4 x86_64 with a TDM2400E
for analog
trunks (& extensions) plus some Polycom 501 & 601 phones.
I have a problem in that
In article <[EMAIL PROTECTED]>,
John covici <[EMAIL PROTECTED]> wrote:
> Hi. I am having a strange problem when using the 1.4 version of
> asterisk and zaptel. If I call from a pstn line into the asterisk box
> using a phone number which calls the box via sip, then once I am in
> the meetme confe
Dave Bolomey wrote:
I've a problem to communicate from a trunk to a another trunk.
My config is :
PBX 1 with subscribers 100 - 119 (SIP trunk to asterisk)
PBX 2 with subscribers 200 - 219 (SIP trunk to asterisk)
Asterisk with Softphone 300 - 319
I can call from Softphone to PBX 1 and from Softp
Hello,
I see the following descriptions on
HYPERLINK
"http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout"rt
ptimeout = Number : Number of seconds, to wait for RTP traffic before
classify the connection as discontinued. Default 0 (no RTP timeout).
(New in v1.2.x).
HYPERLINK
"ht
Wow what a mess! I can imagine how much easier it would be if the world
adopted a country/area/exchange scheme like in the US with known length.
It must be complicated in Germany just within the country. At least in the
US we know what the length should be so if we don't have that we know the
numbe
The 1.4 and 1.2 are alternately on the same box -- I have to install
1.2 rmmod the zaptel modules delete /usr/lib/asterisk/modules/* and
install the zaptel and asterisk. I use the same procedure to go back
to the 1.4. I know the ivr works because I have to use a ivr menu and
even enter a password
Don't bother. If the version of asterisk the crash ocurred in isn't the latest,
the moderators will close the bug.
-Original Message-
From: Vicky [mailto:[EMAIL PROTECTED]
Sent: Friday, December 22, 2006 6:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [as
Rilawich Ango wrote:
you mean we need to remove astdb manual? Totally restart asterisk
even the whole server doesn't do the removement?
To clear the database is one action, yes. You can also use the database
del option to clear entries, but you asked how to start fresh.
As for you question
The + sign is grammatic only it just means your international dialing prefix
"+" the country code etc.
So for dialing a number from Canada to the UK you would advertize the
number as + 44 xx etc. In Canada we dial "011" for international
calls so I would actually dial 01144 xxx
In article <[EMAIL PROTECTED]>,
John covici <[EMAIL PROTECTED]> wrote:
> The 1.4 and 1.2 are alternately on the same box -- I have to install
> 1.2 rmmod the zaptel modules delete /usr/lib/asterisk/modules/* and
> install the zaptel and asterisk. I use the same procedure to go back
> to the 1.4.
LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o
Building modules, stage 2.
MODPOST
*** Warning: "register_wanec_iface"
[/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined!
*** Warning: "unregister_wanec_iface"
[/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined!
CC /r
Looks like you do not have Zaptel source installed, or WanPipe can't
find the Zaptel source.
Colin Anderson wrote:
LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o
Building modules, stage 2.
MODPOST
*** Warning: "register_wanec_iface"
[/root/wanpipe/patches/kdrivers/wanec/wanec.ko] un
Hi
can you help me to build a asterisk manager command event to join a conference?
i've seen that there is the event
Event: MeetmeJoin
Channel:
Uniqueid:
Meetme:
Usernum:
Can you explain me how it works?
Can i use it to join an existing conference?
Thanks
_
Thanks for replying. It's there, /usr/src/zaptel and it's valid. The script
checks for its existence before compiling. The only thing that's unusual
about this zaptel is that it is patched to support KB2 echo can.
Gavin: Thanks, already cc'd Sangoma support. Hope they are working today!
-O
Seems these two are closer to what's asked for.
http://www.vikingelectronics.com/products/view_product.php?pid=343
http://www.vikingelectronics.com/products/view_product.php?pid=217
- Original Message -
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial
In article <[EMAIL PROTECTED]>,
nik600 <[EMAIL PROTECTED]> wrote:
> Hi
>
> can you help me to build a asterisk manager command event to join a
> conference?
>
> i've seen that there is the event
>
> Event: MeetmeJoin
> Channel:
> Uniqueid:
> Meetme:
> Usernum:
>
> Can you e
I think I mixed up when I read AstManProxy files.
So, it does work with 1.4 beta.
That's fine and thanks for the tip
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OK, this brings up a possible lack of understanding by me regarding
the dtmf and sip relationship. I see a dtmfmode in the sip.conf and
it says the mode for sending dtmf -- rfc2833 info or inband . Now
what I don't understand is what controls what asterisk is looking for
in terms of dtmf -- does
On Fri, 2006-12-22 at 05:43 -0400, Humberto Figuera wrote:
> Hi Carlos,
>
> please delete the line "dchan=16".
>
> dchan is use for ccs framing.
>
> You can use the follow:
>
> #
> span=1,0,0,cas,hdb3
> cas=1-15:1101
> cas=17-31:1101
> #
>
Muchas gracias! Every
Does anyone know what the time values in amd.conf are? Are they seconds,
fractions of seconds, heartbeats, what?
;'initialSilence' is the maximum silence duration before the greeting
initial_silence = 25; Maximum silence duration before the greeting.
It doesn't say in amd.conf or at
http:/
how to recovery password spa3000?
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Sangoma support did something and the driver is there. Rebooting now with
the new card. Hold me, I'm scared.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, December 22, 2006 10:09 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE
> Does anyone know what the time values in amd.conf are? Are they
seconds,
> fractions of seconds, heartbeats, what?
Milliseconds.
>
> ;'initialSilence' is the maximum silence duration before the greeting
> initial_silence = 25; Maximum silence duration before the
greeting.
>
> It doesn't
Hi everyone
I just ordered a Sangoma A20001 with 2FXO ports - Does anyone have
suggested reading pointers for what I'll need to do to get it
working? I've only used VoIP in the past so don't know much about
Sangoma drivers or Zaptel. I opted for the non-echo canceling card
so I may need
Hi all,
I'm trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn't see that say, extension 600 is invalid, it just goes back to the
beginning of the
Actually what will happen is they will get REALLY mad that you tried to
blame "your inability to correctly configure a system" on them, then they
will ask for an appology and when you say "why do you deserve one" they will
terminate your account. The incident I had with them involved their site
th
Unfortunately I don't think there is a way to just do a password
recovery. You can, however, do a factory reset. That would reset the
username and password to factory defaults as well as all other settings.
To perform a factory reset you need to access the IVR menu via a
telephone connect t
Hello,
We've released another update to our astGUIclient suite: 2.0.2
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's functionality and the
V
"Andrew Joakimsen" <[EMAIL PROTECTED]> wrote:
> NuFone isnt bad if you want a disposable termination account. But don't rely
> on it for anything.
>
Well, the voice quality left a lot to be desired, so I didn't make a
lot of use of the "service" anyway. Perhaps, if I had made more use of
it, they
Sangoma's website and Wiki has pretty good instructions for the A200
series, even for those of us who aren't fluent in Linux
On the outside chance you venture into the world of pulse dial , you may
need a different driver. Certainly for the FXS side you did. Can't say
if that has been updated la
If this (or any) company is really stealing or not living up to a contract
then why not report them as such, especially if they are US based. I would
suspect you would have another route to take.
If you don't do anything about it then they will just go on abusing
others and getting away with it.
Carlos: Had you tried re-compiling wanpipe?
Had a similar problem, and eventhough I'm pretty sure I compiled
wanpipe and it did the zaptel patch succesfully, once I finished with
the Unicall installation, somehow the patch was not wotking correctly.
So after a couple of days of lo
Hi all,
I'm using 'show translation' to help dimension my system, but I confused
by the results I get. My 2 test systems (results below): an AthlonXP
2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar
results (D930 is slightly faster). Googling shows that others have
simila
Doug Crompton <[EMAIL PROTECTED]> wrote:
> If this (or any) company is really stealing or not living up to a contract
> then why not report them as such, especially if they are US based. I would
> suspect you would have another route to take.
>
I am UK-based - there's not a lot I can do to a US-ba
Well if this company is US based I would not think where you are matters
if it is fraud. You could still enter a complaint at the FBI site.
I would also think they would be working with the UK counterpart.
Doug
On Sat, 23 Dec 2006, Kevin Walsh wrote:
> Doug Crompton <[EMAIL PROTECTED]> wrote:
>
Doug Crompton <[EMAIL PROTECTED]> lazily top-posted:
> Well if this company is US based I would not think where you are matters
> if it is fraud. You could still enter a complaint at the FBI site.
>
> I would also think they would be working with the UK counterpart.
>
I doubt it - not on a minor
Hello
* * * * * * * *
* * * * * Happy X-mas and Adv Happy New Year ...
**
*
--
Thanks and Regards
Ravi Prakash Sunkara
[EMAIL PROTECTED]
M:+91 9985077535
O:+91 40 23114549
F:+91 40 40208727
[EMAIL PROTECTED]
www.hyperion
Leo Ann Boon wrote:
Hi all,
I'm using 'show translation' to help dimension my system, but I confused
by the results I get. My 2 test systems (results below): an AthlonXP
2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar
results (D930 is slightly faster). Googling shows tha
Doug Crompton wrote:
At the very least the BBB (bbb.org) should be notified. They have a web site
SORRY, BUT IMHO the BBB is a joke. I wouldn't waste the time to type in
one word
and if it is really wire/internet fraud then the FBI
(www.fbi.gov/majcases/fraud/internetschemes.htm) has a site y
Hello, I am a Asterisk novice, I am stumped and hope someone can offer me
assistance. If you can, I am willing to pay a consulting fee for such
assistance.
1) I am using Asterisk v1.2.13 for my IP PBX. I am utilizing thinkbright
thinkCarrier 5 SIP (G711) service, utilizing a B2BUA method for
I agree, also have had this religious argument with an international
telephony professional who disagrees
He doesn't see that the variable length numbering (lack of ) plan can
really become a problem
On a similar note, since the NANP has become all electronic, there
should never have been anothe
Hi ALL,
** I like very to desire you and your family, a Merry Christmas, with much
love, peace, professional and personal success.
Best Regards
Josue
2006/12/23, raviprakash sunkara <[EMAIL PROTECTED]>:
Hello
* * * * * * * *
* * * * * Happy X-mas and Adv Happy New Year ...
*
If you have a "provisioned" SPA then you may need to keep it off
the internet after you reset it. I have seen many that go through a
factory reset then "phone home" to get their configuration straight
away from their original supplier. Which of course will lock you out
again ;)
--
Regards,
After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of
which do not have "phone company" compatible FSK/stutter MWI, I finally
got smart and found out just which Panasonic phones have this feature.
Only the following 5.8G models in their current line have FXO compatible
MWI. I
Hi All,
Wish you a very HAPPY and Merry Christmas to all and your beloved once.
Arun
On 12/23/06, Josué Conti <[EMAIL PROTECTED]> wrote:
Hi ALL,
** I like very to desire you and your family, a Merry Christmas, with much
love, peace, professional and personal success.
Best Regards
Josue
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