2011/9/23 amit anand
> Hi
>
> you cannt do itin default CDR of the asterisk, To do so you can use Mysql
> and do it from dialplan
>
How ?
Shall I simply append a new record into MySQL CDR table ?
>
>
> On Thu, Sep 22, 2011 at 18:53, Olivier wrote:
>
>> Dial(SIP/foo,15);
>> if (${DIALSTATUS}="N
I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been
reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy
should be used to provide an interface for Asterisk to get kernel
timing. - espescially if using timing-dependant modules.
I have a minor question: is dahdi_
Hi
you cannt do itin default CDR of the asterisk, To do so you can use Mysql
and do it from dialplan
On Thu, Sep 22, 2011 at 18:53, Olivier wrote:
> Dial(SIP/foo,15);
> if (${DIALSTATUS}="NOANSWER")
>Dial(SIP/bar,15);
>
--
Amit Anand
+91 9818559898
--
Can you please post:
1. Relevant sip.conf
2. sip debug when trying to make a call?
On Thu, Sep 22, 2011 at 7:26 PM, David Björkevik wrote:
> Dear list,
>
> We are switching to a new provider for SIP-trunks. We have 20 numbers,
> each defined as a separate SIP peer.
>
> With the old provider every
Dear list,
We are switching to a new provider for SIP-trunks. We have 20 numbers,
each defined as a separate SIP peer.
With the old provider everything works.
When switching to the new provider's account data, it only works as long
as I only define one of the accounts. If multiple accounts are
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luke Hamburg
Sent: Thursday, September 22, 2011 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bounty for ASTERISK-17474 streaming MusicOnHold
bug
Hi all-
This
Hi rcswebb,
I had a problem like yours :
Asterisk -NAT - internet - NAT - 3CX phone
Without modifiyng Astrisk conf I could start a call from the client but
without hearing a sound.
The solution for me was to force Asterisk to modify the outgoing udp packet
to insert it's public ip and not the p
Hi all-
This is my first post to this list so please don't flog me if this is not
the appropriate place to post this. I've had an issue for over a year
affecting MOH + DAHDI timers, I reported it at:
https://issues.asterisk.org/jira/browse/ASTERISK-17474
Basically the MOH channel goes dead a
On 09/22/2011 11:29 AM, Ian Pilcher wrote:
On 09/21/2011 09:07 PM, Nasir Iqbal wrote:
You can use ictfax HTTP://www.ictfax.org web interface to send faxes,
Ictfax is pure foip software based on t.38 as compared to hylafax
I took a look at ictfax. It's massive overkill for what I need, and the
On 09/20/2011 09:57 PM, Jeff LaCoursiere wrote:
Like most faxing issues, at it's root it is a timing problem IMO.
Sangoma makes a special timing cable to link their cards so you can do
exactly what you are asking to do. I've never purchased it, but last I
looked into the issue, that is what the
On 09/21/2011 09:07 PM, Nasir Iqbal wrote:
> You can use ictfax HTTP://www.ictfax.org web interface to send faxes,
> Ictfax is pure foip software based on t.38 as compared to hylafax
I took a look at ictfax. It's massive overkill for what I need, and the
setup looks far from simple. (And that's
September 22, 2011 11:20 AM Kevin P. Fleming wrote:
>For many people, with modern CPUs, current versions of DAHDI and Asterisk,
and appropriate configuration (using the faxbuffers option in
chan_dahdi.conf, for example), such a system can be setup to work very, very
close to 100% of the time.
T
On 09/22/2011 02:20 AM, Olivier wrote:
>
> Doesn't Zoiper include some T.38 features ?
>
It does, but the free version adds watermarks. It also doesn't appear
to have been packaged for recent versions of Fedora (or any 64-bit
version).
--
==
On 09/20/2011 03:43 PM, Adam Moffett wrote:
If I have a 4 port Digium FXS card and a single port PRI card on the
same asterisk box, is it expected that I'd be able to plug a fax machine
into the analog FXS port and have no problems sending or receiving
faxes? Our connection to the Telco is on the
On 09/20/2011 08:57 PM, Don Kelly wrote:
This is a scary answer—you’re saying that what should be simple “TDM”
FXS to PRI does not work?
There is no TDM connection on a PCI or PCI-Express bus. Transferring
data between two cards in the system either requires a direct connection
between them
Sounds like a great idea.. Hopefully the page/account never gets hacked and
bad IP's published.. I could see a great hack of
127.0.0.1
192.168.0.0/16
10.0.0.0/8
getting up there somehow and next thing you know - BAM!
But I haven't RTFM - I'm guessing there is probably a white list that
very cool!
On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo wrote:
>
> Apologies for cross posting but some of us aren't on the other list
> (vice/versa) and thought both groups would benefit.
>
> For those familiar with the VoIP Abuse Project, no need to explain the
> gist of this. I got tired of pa
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long an
Hi look at option a. This option put on accountcode field the name on the left
your password file.
Regards
Enviado desde mi iPad
El 22/09/2011, a las 5:49, Malvin Rito escribió:
> Hi,
>
> I tried Authenticate where pass codes are stored on the file pass.conf and it
> works.
>
> exten => _
Hello,
In my 1.6.1.18-powered system, I've got the following dialplan (in
extensions.ael) :
Dial(SIP/foo,15);
if (${DIALSTATUS}="NOANSWER")
Dial(SIP/bar,15);
When SIP/baz dials peer SIP/foo which do not answer, I've got a single CDR
entry like this:
SIP/baz SIP/bar time_when_foo_sta
Dear Malvin,
I see Sam worked hard to post you the whole info about the application where
it clearly states the use of option "a" - Please change the configuration
line accordingly now and see if it works for you.
Best Regards,
Gohar
From: asterisk-users-boun...@lists.digium.com
[mailto:ast
Hi,
I tried Authenticate where pass codes are stored on the file pass.conf
and it works.
exten => _,1,Authenticate(/etc/asterisk/pass.conf)
Since I have my CDR, I want to have a report wherein I can check which
pass code did the call. How can I achieve it?
Using authenticate thro
Yes, Zoip support T.38 faxing but It is only client application and you need
FOIP gateway (asterisk) to transmit a fax to your FXO port
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Thu, Sep 22, 2011 at 3:20 AM, Olivier wrote:
>
>
> 2011/9/21 Ian Pilcher
>
>> I am looking f
2011/9/21 Ian Pilcher
> I am looking for a simple way to send occasional faxes via the FXO
> port on my SPA3102 -- without having to connect a fax modem to an
> ATA. In an ideal world, this would be some sort of "softfax" that
> runs on my Linux desktop and talks (via Asterisk) to the SPA3102 wi
24 matches
Mail list logo
