Hi,
I am using ubuntu-12.04 and installed asterisk from repository (apt-get
install asterisk).
I have configured it to work with odbc,
*CLI odbc show
ODBC DSN Settings
-
Name: asterisk
DSN:asterisk-connector
Last connection attempt: 1970-01-01 05:30:00
Thanks a lot Gareth,
I just took the astdb file from an older version of the same system and it
worked after a few adaptation (registers, and some DB puts/del).
Regards,
Samuel.
On 30 July 2013 11:30, Gareth Blades mailinglist+aster...@dns99.co.ukwrote:
On 29/07/13 18:12, samuel wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to passtrough T38 Faxes with any asterisk
version at all?
What do you mean? Am I able to
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e71378 is ringing
-- SIP/228-09e71378 answered Zap/26-1
== Spawn
* PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE *
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10)
A J Stiles wrote:
* PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE *
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1]
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to passtrough T38 Faxes with any asterisk
version at all?
What do
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
Most of my experience until recently has been in Asterisk 1.2, and I am
just starting to make use of Asterisk 11 for new systems.
I have a question about using SIP on a multi-homed machine.
I have a customer who wants an Asterisk box with two network interfaces:
one on the public Internet (no
On 7/31/2013 10:32 AM, Tony Mountifield wrote:
Most of my experience until recently has been in Asterisk 1.2, and I am
just starting to make use of Asterisk 11 for new systems.
I have a question about using SIP on a multi-homed machine.
I have a customer who wants an Asterisk box with two
This is the standard way we set up our servers. There is nothing special
about it. Just make sure you disable direct media.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield
Sent:
Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten =
On 31/07/13 15:32, Tony Mountifield wrote:
Most of my experience until recently has been in Asterisk 1.2, and I am
just starting to make use of Asterisk 11 for new systems.
I have a question about using SIP on a multi-homed machine.
I have a customer who wants an Asterisk box with two network
In article 51f925f2.1040...@dns99.co.uk,
Gareth Blades mailinglist+aster...@dns99.co.uk wrote:
On 31/07/13 15:32, Tony Mountifield wrote:
Most of my experience until recently has been in Asterisk 1.2, and I am
just starting to make use of Asterisk 11 for new systems.
I have a question
In article
616B4ECE1290D441AD56124FEBB03D08171492B22C@mailserver2007.nyigc.globe,
Eric Wieling ewiel...@nyigc.com wrote:
This is the standard way we set up our servers. There is nothing special
about it. Just make sure you disable direct media.
Thanks, that's reassuring. Appreciate the
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten =
On 31/07/13 16:12, Tony Mountifield wrote:
Thanks. But I thought localnet= and externip= were for when the external
interface is going through NAT. In this case the ITSP is connected through
a real non-NATted public interface.
Is it possible to specify directmedia=no just for the SIP trunk? So
On 13-07-29 10:22 AM, Eduardo Leones wrote:
Hello, working in a call center where we set up a structure in asterisk.
When my voip reaches 150 calls are with bad quality. We do not transcode
codec. What I realized using the top command server (CentOS) processing is
too high for the asterisk. But
Here's how I set up Asterisk in my SOHO installations. For most of these,
the asterisk box is dual homed and some are also the site's
router/firewall. The config is the same either way.
sip.conf
[general]
bindaddr = 0.0.0.0:5060
externhost = something ; for the sites with a dynamic ip.
externip
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to passtrough T38
Larry Moore wrote:
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to
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