Hi,
I was wondering if anyone you got asterisk to work with Fax via google voice
? If so, can you please send me extension.conf and sip.conf, jabber.conf
and gtalk.conf settings used. I would prefer faxing with Fax for Asterisk
(FFA) via .call file. I see post where people got it work with Googl
Dear wcselby;
Thanks a lot for your reply.
For below script, I have some questions if you can help me:
1) I am looking to have reports for the call center, so I need to determine how
many calls in the queue, and how many agents logged and when the agent logged
in and when logged out ... etc. B
I have been sending OPTIONS requests both programatically (my own code),
manually via SIP VERIFY PEER x and automatcially by setting verify=yes in
sip.conf. The trouble is I do not see anything except an ACK 200 come back
from endpoints and it does not contain any SDP/codec info. . My goal is to
de
Hi All;
I am using asterisk version 1.8 and I selected CDR mysql from the menuselect
when I was doing the compilation and installation.
How can I know if the unique id will be added to the cdr, and how I can know
which information will be logged, also from where I can new field to be logged
an
On Tue, Oct 25, 2011 at 7:30 AM, bilal ghayyad wrote:
> Dear Tark;
>
> The asterisk version I am running is 1.8 and I can select mysql from
> menuselect when I am compiling.
>
> But when I googled for cdr-mysql, I discovered that I have to login for
> mysql and create the database and run a scrip
On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger wrote:
> Customer 200 calls to queue 900, Agent 300 answers but tells Customer 200
> that he should be at Queue 901 and transfers Customer 200 (using *2) to
> Queue 901. Agent 301 now gets the call from Queue 901 with Customer 200,
> answers the calls
On Fri, Oct 21, 2011 at 1:15 PM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:
> hi
> here is my extensions.conf and aheeva_diaplan.conf
>
> if you can see theses files and tell my if there is any wrong
>
> regards
>
>
>
These configuration files you sent me don't seem to match up w
Le 25/10/2011 13:30, Yaroslav Panych a écrit :
Hello
Always returns 401 Unauthorized, because of
[Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on
stale nonce received from '"L6"
Change the local port from the DLInk (eg 5060 to 15060) and it should
work. After few hours yo
2011/10/25 Tarek Sawah :
> Hello,
> Is L6 a remote device? is there any firewall residing between the server and
> UA?
>
>
> Tarek Sawah
>
> Information Technology Adviser
>
> Integrated Digital Systems
>
> CCNP, MCSE, RHCE, TELECOM
>
> USA: +1 386 492 9993
>
L6 is account of DLINK DVG7022S VoIP
That's the ticket - I now feel justified in my reasoning for the 10.0
upgrade (no re-training for my receptionist!)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller
Sent: Thursday, October 20, 2011 1
In my 1.4 and 10.0 experience, there are two types of variables. Call/Local
variables that "live" for the duration of the current call only and Global
variables that are set and live on until Asterisk is restarted. I can do
set(foo=${CALLERID(num)) and use ${foo} anywhere in my dialplan. I can
We used Mix of Nagios, Zabbix, OpenNMS. Best one for this was Zabbix.
On Tue, Oct 25, 2011 at 6:49 PM, Ishfaq Malik wrote:
> Which monitoring tool were you using?
>
> On Tue, 2011-10-25 at 18:46 +0500, Sammy Govind wrote:
> > I wrote my own shell scripts to collect "core show calls" value from
>
Which monitoring tool were you using?
On Tue, 2011-10-25 at 18:46 +0500, Sammy Govind wrote:
> I wrote my own shell scripts to collect "core show calls" value from
> asterisk and then push the filtered value to an opensource monitoring
> tool. That worked perfectly well.
>
>
> #!/usr/bin/perl -w
i like asterisk -rx 'show channels concise'
give less detailed but more readable output
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: da...@debsinc.com
> To: asterisk-users@lists.digium.com
> Date: Tue, 25 Oc
I wrote my own shell scripts to collect "core show calls" value from
asterisk and then push the filtered value to an opensource monitoring tool.
That worked perfectly well.
#!/usr/bin/perl -w
use strict;
open(LINE, 'asterisk -rx "core show channels"|');
my ($chans, $calls, $line)=(0,0,undef);
whil
The "Simplest" method of seeing the number of concurrent calls is "service
asterisk status". If I understand question two, asterisk -rx " core show
channels verbose" is probably your best bet.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...
Hi
What are people using to monitor the concurrent number of calls at any
given time?
Also, is there any good way of monitoring concurrent inbound and
outbound calls so that we can see the 2 different numbers?
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161
Hello,
Is L6 a remote device? is there any firewall residing between the server and UA?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: panyc...@gmail.com
> Date: Tue, 25 Oct 2011 14:30:53 +0300
> To: asterisk-
Dear Tark;
The asterisk version I am running is 1.8 and I can select mysql from menuselect
when I am compiling.
But when I googled for cdr-mysql, I discovered that I have to login for mysql
and create the database and run a script to create this and give the grants.
All what I found in google
Hi all,
We are encountering some problems with the queuemanager in some specific
cases.
When an agent gets a call from a queue the agent gets a wrap-up time
after the call is finished, so far so good.
When the call enters the queue while being transferred from an agent
from another queue ther
Hello
Always returns 401 Unauthorized, because of
[Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on
stale nonce received from '"L6"
;tag=31b9dc9e-684902'
L6 is realtime device of type FRIEND (DLINK DVG7022S)
Reviewed SIP conversation - no results.
SIP debug
<--- SIP read fro
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