Is there any way to ring the PC speaker instead of sound card when calling
Console?
Yuan Liu
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What determines timeout for applications/functions like ENUMLookup,
DUNDiLookup? Any way to modify?
Yuan Liu
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Hi, can i set up a similar dialplan?
Suppose that extension make the caller to join a conference room.
extension => 3,1,Dial(SIP/200,SIP/[EMAIL PROTECTED])
extension => 3,2,Dial(SIP/201,SIP/[EMAIL PROTECTED])
extension => 3,3,Dial(SIP/[EMAIL PROTECTED])
After that, the caller, the user 200
Le 13 janv. 07 à 02:10, Leo Ann Boon a écrit :
Antoine Fressancourt wrote:
Hello,
Thank you Leo for your answer,
I manage to do what I want perfectly when both the caller and the
callee are set in SIP with canreinvite=no using SIP INFO method
for DTMF.
Now, I can't figure out why this
On Sat, 13 Jan 2007, John French wrote:
I'm trying to learn to use fxotune and am getting the error: Could not
fill input buffer - got -1 bytes, expected 4000 bytes Failure! when I
run it. This is a new install of Asterisk 1.4 and zaptel 1.4. Also I
can't really find an up to date how to on
I think this is wrong use,
for conferencing, you must dial meetme application and put all users
into one conference room...
PJ
nik600 wrote:
Hi, can i set up a similar dialplan?
Suppose that extension make the caller to join a conference room.
extension => 3,1,Dial(SIP/200,SIP/[EMAIL P
On 1/14/07, Pavel Jezek <[EMAIL PROTECTED]> wrote:
I think this is wrong use,
for conferencing, you must dial meetme application and put all users
into one conference room...
PJ
don't worry, meetme application is called on extension
I've written this code using phpagi:
connect()) {
The easy answer is to use the "r" switch with Queue, still you may want to
use a MOH that "fakes" a ringing tone in order to have audio messages
smoothly mixed in :)
l.
On Thu, 11 Jan 2007 23:33:40 +0100, Ex Vitorino <[EMAIL PROTECTED]>
wrote:
Hello List,
This must be an easy one
*Olá pessoal,
alguém pode me ajudar, como posso solucionar isto?
toda hora que preciso parar um iniciar um serviço me mostra isto, e tenho
alguns serviços que não esta rodando corretamente, pelo que eu estudei é
relacionado ao numero de processo, não ainda não achei uma solução.
/etc/init.d/fun
Chris Bullock wrote:
Hi. I've been researching very deep into SLA in Asterisk 1.4, and am unclear
as to if the feature exist. I know the commands and configs are available,
but there is no documentation, and I've heard that it will not be supported
until 1.4.1. Does anyone have a definitive answ
Al wrote:
I'm facing a weird issue, polycom phones work fine in the main office,
in remote office it says,
Registration from '' failed for '70.59.21.112' -
Wrong password
the odd thing is Linksys phone works without any issue!!
Just a guess, put nat=yes in the sip.conf for that phone and see
I have never seen a registration failure solved with nat=yes.
Doug Lytle wrote:
Al wrote:
I'm facing a weird issue, polycom phones work fine in the main office,
in remote office it says,
Registration from '' failed for '70.59.21.112' -
Wrong password
the odd thing is Linksys phone works witho
Are you using tftp or ftp provisioning? If so, check your server
declaration in sip.cfg in your polycom configs directory.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.shelto
Eric, I did, what's happening is that the unauthorized message from
the registering Polycom never reaches the Polycom because asterisk
doesn't know it's natted. Therfore the Polycom never supplies the
creditentials.
On 1/14/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
I have never seen
Eric "ManxPower" Wieling wrote:
I have never seen a registration failure solved with nat=yes.
Just a guess, put nat=yes in the sip.conf for that phone and see if
it helps.
Hence the, "Just a guess".
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a littl
Hi Francois,
Thank you for your interest.
I tried the card alone so I don't think is an IRQ problem (anyway is there a
way to be sure?)
To be sincere, in all systems I saw (even working ones) there is an IRQ
balance failed where Linux boots.
The system is new, I tried different process
On Sun, 14 Jan 2007, Giuffredi wrote:
I tried the card alone so I don't think is an IRQ problem (anyway is
there a way to be sure?)
You can make sure the card is sitting on it's own IRQ - use the command
cat /proc/interrupts
to check. Eg. on one of my boxes with a TDM400P:
CPU0
nat is equal to yes,
and server definition in ftp provisioning server is correct.
i followed packets between phone and asterisk,
it seems for some reason asterisk is not happy about challenge response its
getting from polycom.
why its not happening in the same LAN, beats me!
and also NAT device is
Hola
Que distribucion usas?
On 1/14/07, Jugleni Jr <[EMAIL PROTECTED]> wrote:
*Olá pessoal,
alguém pode me ajudar, como posso solucionar isto?
toda hora que preciso parar um iniciar um serviço me mostra isto, e tenho
alguns serviços que não esta rodando corretamente, pelo que eu estudei é
Is your Cisco device a Cisco router if so make sure you have no sip
fixup.
The Cisco may be fudging the SIP headers.
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AL Daei
Sent: Sunday, January 14, 2007 11:57 AM
To: asterisk-user
I finally isolated the problem, hope it works for you too. The default -m
(silencegoodfor) default of 18 seconds is too long for my telco and the test is
getting interrupted. I listened in on the line with a splitter and realized
what was happening. I had to set -m down to 15 seconds in my case
So, do the same setup with the DNS, setup a colocated machine and your
asterisk inside the LAN registers with it, in your dialplan instead of:
exten => 123,1,Dial(SIP/123,25,r)
exten => 123,2,VoiceMail(u123)
Do:
exten => 123,1,Dial(SIP/123,25,r)
exten => 123,2,Dial(IAX2/colocation-asterisk/${ex
What video clip? Does a native video call between the two work?
On 1/14/07, Antoine Fressancourt <[EMAIL PROTECTED]> wrote:
Le 13 janv. 07 à 02:10, Leo Ann Boon a écrit :
> Antoine Fressancourt wrote:
>> Hello,
>>
>> Thank you Leo for your answer,
>>
>> I manage to do what I want perfectly wh
Hi list,
I've installed Asterisk 1.4.0 with newest mISDN 1.0.4 + mISDNuser 1.0.3
on Fedora Core 6.
I get many compilation error on mISDN. It wants to include linux/config.h
That I fixed by removing the #include line at every occurance. (Don't
know if that was a wise move, but it then compiled).
Hi All,
I'm testing different ways to implement a LCR/OCN tabe to shift calls
to multiple carriers for better rates. I'm using realtime mysql
database access, asterisk 1.2.9.1 with mysql 3.23.
Scenario 1:
I send outgoing calls with with a Goto statement into the context with
the realtime switch
CentOS
2007/1/14, Carlos Rojas <[EMAIL PROTECTED]>:
Hola
Que distribucion usas?
On 1/14/07, Jugleni Jr <[EMAIL PROTECTED]> wrote:
> *Olá pessoal,
>
> alguém pode me ajudar, como posso solucionar isto?
>
> toda hora que preciso parar um iniciar um serviço me mostra isto, e
> tenho alguns se
Hi there.
Perhaps my message lost meaning when I post the answer to Eric's
question without stating the problem again...
Is there a way I can tell asterisk to play *vm-isunavail* (person
unavailable) instead of *vm-isonphone* (person is on the phone) when DND
is enabled on one of the extenst
I have noticed that Asterisk (version 1.2.13) is not hanging up a call
when the wifi handset moves out of range.
My setup is Nokia E61 connected to wifi access point (private IP range)
and then to server on internet (public IP).
I have been testing using the talking clock application, and walking
I don't have a particular reason to upgrade, but I'm installing a new box,
so I have the opportunity to go 1.4. On the other hand, I'm not familiar
with 1.4, and relatively new to Asterisk. So instead of trying to keep up
with two different versions, I want to tie my handful of boxes to one,
On 10:13, Sat 13 Jan 07, Pierre du Plessis wrote:
> Hi there.
>
> Perhaps my message lost meaning when I post the answer to Eric's
> question without stating the problem again...
>
> Is there a way I can tell asterisk to play *vm-isunavail* (person
> unavailable) instead of *vm-isonphone* (pers
When I send a call from my TE410P using E&M, the legacy PBX answers
the call but doesn't route it. Any idea what this could be? I assume the
digits aren't being delivered properly to the legacy pbx. Any
suggestions on what config settings to muck with?
Asterisk SVN-branch-1.2-r40901 built b
Change log can help you a lot. I would stick to my grandmothers
advice, "if it aint broken don't fix it".
On 1/14/07, Yuan LIU <[EMAIL PROTECTED]> wrote:
I don't have a particular reason to upgrade, but I'm installing a new box,
so I have the opportunity to go 1.4. On the other hand, I'm not fa
cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
just fine. (to make matters worse, it does seem to work sometimes,
although once working or not working between changes it either works or
doesn't work all the time.)
extensions.conf:
[general]
static=yes
writeprotect=no
autof
Am Sonntag, den 14.01.2007, 17:13 -0800 schrieb chester c young:
> cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
> just fine. (to make matters worse, it does seem to work sometimes,
> although once working or not working between changes it either works or
> doesn't work all
--- Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
> Am Sonntag, den 14.01.2007, 17:13 -0800 schrieb chester c young:
> > cannot make Dial(...,,g) work correctly, although Dial(...,,gh)
> works
> > just fine. (to make matters worse, it does seem to work sometimes,
> > although once working
I am currently running Zaptel 1.4 with Asterisk 1.2.
From: [EMAIL PROTECTED] on behalf of Trevor Peirce
Sent: Fri 1/12/2007 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] zaptel & asterisk versions (was Echo...)
Am Sonntag, den 14.01.2007, 17:13 -0800 schrieb chester c young:
> cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
> just fine. (to make matters worse, it does seem to work sometimes,
> although once working or not working between changes it either works or
> doesn't work all
I've been through this once and, IIRC, I'd say your [outside]
context in the dialplan does not have an extension matching
the incoming msn.
Suggestion:
- Recheck your extensions.conf [outside] context
Otherwise:
- Increase misdn debug to level 3
(something like "misdn set debug 3" a
--- Gordon Henderson <[EMAIL PROTECTED]> wrote:
> On Sun, 14 Jan 2007, Giuffredi wrote:
>
> > I tried the card alone so I don't think is an IRQ
> problem (anyway is
> > there a way to be sure?)
>
> You can make sure the card is sitting on it's own IRQ -
> use the command
>
>cat /proc/inte
Hello,
I am looking to purchase a new quad-band cellphone and I'm looking for
one with WiFi and enough CPU power for stable SIP calls. I was wondering
if anyone here can share his experience and recommend on a good
cellphone. Any chance there is such a phone with even good WiFi profiles
manag
Hey Tomer,
I'm not sure if the Audiovox PPC6700 is quad band, but it does support Wifi and
runnings SJPhone great! It is even usable over Sprints EVDO service.
On Mon, 15 Jan 2007 08:01:44 +0200, Tomer Horn <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I am looking to purchase a new quad-band cellphon
> in my case I want a user to be on-line all the time - the system will
> dial and connect them and, when they're done, they select the next one.
> what I'm doing now is putting them into a loop with a g-option on the
> dial. the number it dials is set thru the api. if the number's not
> set it
Perhaps MusicOnHold() app?
> Is there any application can let the dial plan to execute while
> playing music? Say I have a lot of command to do in the dial plan but
> I don't want to keep silence while execution of dial plan. I notice
> Background(file) can play music but it will hold until pre
It doesn't work as it will hold up the call without running the rest
of the statement.
On 1/12/07, Octavio Ruiz (Ta^3) <[EMAIL PROTECTED]> wrote:
Perhaps MusicOnHold() app?
> Is there any application can let the dial plan to execute while
> playing music? Say I have a lot of command to do in
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