Hi
Greeting to all you out there.
I am new at asterisk, I have been working with PLMN platforms telecommunication
for 5 years with NSN and Huawei.
We have recently built an asterisk PBX with Trixbox and connected it to our MSS
using Digium E1 cards(ISDN). Everything went smoothly as there are
On Wed, Sep 25, 2013 at 3:22 AM, Endri Stefani endri.stef...@plus.alwrote:
Hi
** **
Greeting to all you out there.
** **
I am new at asterisk, I have been working with PLMN platforms
telecommunication for 5 years with NSN and Huawei.
We have recently built an asterisk PBX
stack
[2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing
[s@sub-record-check:17] Set(SIP/1002-0292, __YEAR=2013) in new stack
[2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing
[s@sub-record-check:18] Set(SIP/1002-0292, __TIMESTR=20130925-135733)
in new stack
[2013-09-25 13
It would be easier to comment if you would copy the relevant part of the
dialplan.
The only dial cmd I see is
Dial(SIP/1002-0292, DAHDI/g1/99,300,wW)
At least where I live, 99 can be part of a valid telephone number. You could increase
the verbosity and see what Dial() is
On 25/09/13 09:54, Kumar Shantanu wrote:
I am facing a strange problem on my asterisk box (using isdn lines
with pri card installed on it). Normal incoming/outgoing calls are
working perfectly fine.
When a user dial a wrong/out-of-service number they don't hear back
any such message like The
Thanks Gareth ,
Try calling Progress() just before the dial command. Without this
Asterisk wont send the SIP/183 Session Progress and send the inband
audio until the call is answered.
Do I need to change something in asterisk dial plan ? I am using freepbx
to mange asterisk graphically.
It took an OpenVox engineer to sort out this obscure problem in the end, but
it was pretty much as I suspected: the Quectel M20 GSM module serving span 1
was stuck in an unusual state, in which it would only operate in the 900 MHz
band. Fine for O2, Vodafone and Tesco; but no good for
On 09/25/2013 09:22 AM, Endri Stefani wrote:
Hi
Greeting to all you out there.
I am new at asterisk, I have been working with PLMN platforms
telecommunication for 5 years with NSN and Huawei.
We have recently built an asterisk PBX with Trixbox and connected it to
our MSS using Digium E1
On 25/09/13 11:21, Kumar Shantanu wrote:
Thanks Gareth ,
Try calling Progress() just before the dial command. Without this
Asterisk wont send the SIP/183 Session Progress and send the inband
audio until the call is answered.
Do I need to change something in asterisk dial plan ? I am using
Hi. i am running asterisk 11.5.1 in my system (debian squeeze) and i do get
the CDRs through the csv file, that asterisk creates.
i would like to have the CDRs in a nice web based tool and after some
search i have found
http://acdr.com.au/
i do have it installed with all the dependencies (apache,
We can use the Dial() command with the 'r' option in order to generate
the UK ringtone (as we are UK based the default is UK).
How do we generate a USA ringtone for example?
I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us'
(and calling Progress() beforehand) and
Hi Patrick,
If I use latest stable asterisk will I be able to change dialplan by changing
pridialplan in chan_dahdi.conf?
Br
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Wednesday,
On 09/25/2013 01:57 PM, Endri Stefani wrote:
Hi Patrick,
If I use latest stable asterisk will I be able to change dialplan by changing
pridialplan in chan_dahdi.conf?
AFAIK yes.
You may also want to check out Asterisk The Definitive Guide (4th
edition is the latest). Paperback version:
On 2013-09-25 09:22, Endri Stefani wrote:
Hi
Greeting to all you out there.
I am new at asterisk, I have been working with PLMN platforms telecommunication
for 5 years with NSN and Huawei.
We have recently built an asterisk PBX with Trixbox and connected it to our MSS
using Digium E1
if you are a serious teleco guy, which it seems you are. you might consider
dumping trixbox in the near future. while trixbox does provide a good entry
level into the * world, there are limitations that will eventually hold you
back from enjoying the full breadth of utility that * offers.
food
Thank you Gareth,
It worked like a charm.
The only problem I am having is now, when I do some changes in my
freepbx and reload it just rewrites my dial play , I will try to fix it
though.
Thanks again
Cheers
Shantanu
On 09/25/2013 05:09 PM, Gareth Blades wrote:
On 25/09/13 11:21, Kumar
Hi guys
Thanks a lot, I am just getting used to it, my telco managers ☺ don’t trust
stability for open source solution for voice(you give a headache to calling
parties if lag is more than 250ms ☺ ) and I want to prove them wrong. I have
successfully integrated * with our system via ISDN and
but i do not know how to interface the CDRs. has anyone used this tool or
any other similar tool?
It expects your CDR to be located in a mysql database. You'll either need to
figure out how to import your .csv into mysql, or have Asterisk send the CDR
directly to the mysql database.
Doug
i have uploaded two files for you
one is the php script that reads Master.csv and loads data in the database
you have to change
$locale_db_name = 'databasename';
$locale_db_login = 'user';
$locale_db_pass = 'password';
this how you run the script
php /path/cdrmysql.php
I asked you before. What exactly are you trying to do that you cannot? It
helps to be fairly detailed when asking a question to the list. Include
error messages if you have any.
The dialplan and your ISDN configs are different things. It sounds like
maybe you are having issues with with your
chan_dahdi.conf.sample, included in the Asterisk source code, suggests
pridialplan=unknown. The CALLERID() functions has a couple of options for
setting num plan, but the options are not well documented.
; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
; the
Hi Steve,
There are no errors I need to be able to change TON(below my PRI debug ) in
international or subscriber. The change in chan_dahdi.conf did not do it
Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
So you are using QSIG and connecting your Asterisk box to a legacy PBX over
PRI E1?
Did you try unknown?
Do you need to use QSIG (over euroisdn for instance)?
Thanks,
Steve Totaro
On Wed, Sep 25, 2013 at 10:33 AM, Endri Stefani endri.stef...@plus.alwrote:
Hi Steve,
** **
There are
The call is being placed, is it not? Again, I know you are trying to
change the TON but what are you trying to accomplish and what is failing.
It seems like you are dialing 1000 and that is being sent on the wire.
Thanks,
Steve Totaro
On Wed, Sep 25, 2013 at 10:37 AM, Steve Totaro
Hi Guys,
Anyone ever seen this before.
on asterisk 1.8 if i set one of my pabx extensions to show private
number and send a call over VoIP with g729 the call fails but with alaw
it works.
If i enable the callerid on g729 it also works
see error below
From:
On 25/09/13 13:57, Kumar Shantanu wrote:
Thank you Gareth,
It worked like a charm.
The only problem I am having is now, when I do some changes in my
freepbx and reload it just rewrites my dial play , I will try to fix
it though.
Thanks again
I did see in the console output it doing a
On 25/09/13 15:42, Andrew Colin wrote:
Hi Guys,
Anyone ever seen this before.
on asterisk 1.8 if i set one of my pabx extensions to show private
number and send a call over VoIP with g729 the call fails but with
alaw it works.
If i enable the callerid on g729 it also works
see error below
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