Hi
i am not a expert on Asterisk and search a lot of small information :
I use Asterisk 1.6.1.4 with MySQL.
That's work and in my extension.conf, i have:
[as5300-incoming]
switch = Realtime
and in extconfig.conf
extensions = mysql,general,VOIP_Extensions
A lot
[ivr_holiday]
switch = Realtime/ivr_holid...@extensions
where 'ivr_holidays' is context and 'extensions' is table
On 01.10.2010 12:52, Phibee Network Operation Center wrote:
Hi
i am not a expert on Asterisk and search a lot of small information :
I use Asterisk 1.6.1.4 with
On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote:
That's work and in my extension.conf, i have:
[as5300-incoming]
switch = Realtime
and in extconfig.conf
extensions = mysql,general,VOIP_Extensions
A lot of Extension are into the table VOIP_Extensions.
Thanks, it's limited the number of table ?
Le 01/10/2010 11:07, Захаров Антон a écrit :
[ivr_holiday]
switch = Realtime/ivr_holid...@extensions
where 'ivr_holidays' is context and 'extensions' is table
On 01.10.2010 12:52, Phibee Network Operation Center wrote:
Hi
i am not a
Le 01/10/2010 11:10, Steve Howes a écrit :
On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote:
That's work and in my extension.conf, i have:
[as5300-incoming]
switch = Realtime
and in extconfig.conf
extensions = mysql,general,VOIP_Extensions
A lot
Hello,
The harddisk of my etch/bristuffed asterisk1.2 box finally died. I moved
the cheap (1397:2bd0) HFC-S card to a squeeze host (i686) and built
dahdi modules 2.3.0.1 using m-a. After zaptel-dahdi and asterisk
1.2-1.6 config adaptations, everything seems ok, except for the BRI
side, unable to
After test, that's don't work :=
Le 01/10/2010 11:07, Захаров Антон a écrit :
[ivr_holiday]
switch = Realtime/ivr_holid...@extensions
where 'ivr_holidays' is context and 'extensions' is table
On 01.10.2010 12:52, Phibee Network Operation Center wrote:
Hi
i am not a expert on
I am also facing the call disconnection if there is a third call. I tried
disable call waiting in the BRI router, but now it has been reduced, it
means call disconnection is not permanent but seems to be occasion, let say
per day two times there is a call disconnection.
On Wed, Sep 29, 2010 at
What happens if you change to:
signalling=bri_cpe_ptp
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Sent: 01 October 2010 11:37
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Hello,
Perhaps i'm wrong but i don't find a real documentation for cmd Bridge (i've
take a look to the source code but i'm not a guru and i probabely miss
something):
Is it possible as for cmd meetme to have a context to return on 'exit'/end of
the bridge? (in fact i think 'no')
I've done a
Danny Dias wrote:
Hello,
We are having issues with a NEW Sangoma A108D:
-- Executing [691918...@pbx1:1] Dial(SIP/xtravoip200-009d24b0,
DAHDI/g0/691918892|30|m) in new stack
[Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No
translator path exists for channel type
I'm having problems with DTMF on outgoing DAHDI. See
https://issues.asterisk.org/view.php?id=18084
I've tested that the dtmf comes into * correctly. I know that it works
if I use SIP outbound. But it doesn't work if I use DAHDI outbound.
I'd like to figure out what the called party on outbound
When calling Originate Action, it rings my phone. If I do not answer, I
receive a Channel Event: Hangup, followed by receiving an
OriginateResponse event with a Failure Response, Reason 3.
My phone continues to ring.
If I answer the phone at this point, it attempts to answer, but does not
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Friday, October 01, 2010 3:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AMI Originate
snip
Timeout: 3
3 seconds to answer the
3 miliseconds...
2010/10/1 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Cropp
*Sent:* Friday, October 01, 2010 3:50 PM
*To:*
Even Wally can't get coffee (or pick up the phone) that fast!
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang
Sent: Friday, October 01, 2010 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Fri, Oct 1, 2010 at 10:08 AM, Danny Dias ing.diasda...@gmail.com wrote:
Hello,
We are having issues with a NEW Sangoma A108D:
-- Executing [691918...@pbx1:1] Dial(SIP/xtravoip200-009d24b0,
DAHDI/g0/691918892|30|m) in new stack
[Oct 1 10:04:43] WARNING[14171]: channel.c:3170
On Fri, Oct 01, 2010 at 01:49:48PM +0100, Andrew Thomas wrote:
What happens if you change to:
signalling=bri_cpe_ptp
It's bri_cp , not bri_cpe_ptp .
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
unsubscribe
On Oct 1, 2010 8:58 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Fri, Oct 01, 2010 at 01:49:48PM +0100, Andrew Thomas wrote:
What happens if you change to:
signalling=bri_cpe_ptp
It's bri_cp , not bri_cpe_ptp .
--
Tzafrir Cohen
icq#16849755
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