/isdncausecodes.php
I have removed crc4 from my /etc/dahdi/system.conf file, because BSNL in
India not using crc4.
Michael.k
On Wed, Oct 19, 2011 at 10:43 AM, michael k wrote:
> Hi List,
>
>My all incoming calls are working fine but i cant make outgoing
> calls. There w
-- Forwarded message --
From: michael k
Date: Wed, Oct 19, 2011 at 10:43 AM
Subject: Outgoing call failure
To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users@lists.digium.com>
Hi List,
My all incoming calls are working fine but i can
Hi List,
My all incoming calls are working fine but i cant make outgoing
calls. There was no issues for both incoming and outgoing calls till
yesterday. Can somebody tell me what is the issue is ?. I have enable the
dibugging in pri line by issue the command "pri set debug on span 1"
Hi List,
What is the diffidence between A Generic SIP Device and
Generic ZAP Device while we create an extension in FreePBX ?
Mic
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to As
Hi List,
I have configured TE121PF card in E1 mode. I am using asterisk
1.6 and freepbx 2.7. My card staus turn in to green and looks like sync with
the service provider. My service provider is BSNL - India. I have one toll
free number for incoming and one land line number for out goi
as well.
>
>
> On Fri, Oct 7, 2011 at 7:07 PM, michael k wrote:
>
>> Hi,
>>
>>
>> This is my /etc/asterisk/chan_dahdi.conf file.
>>
>>
>> [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf
>> ; Copied from DAHDI Mo
etc/asterisk/chan_dahdi.conf
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
> Sent: Friday, October 07, 2011 9:24 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
Found
Thanks,
Michael.k
On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling wrote:
> What happens when you do the module load chan_dahdi.so command?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Beha
s "module unload chan_dahdi.so" and
> "module load chan_dahdi.so".
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
> Sent: Thursday, October 06, 2011
s-busy-now&pls-try-call-later, noanswer") in new stack
-- Playing 'all-circuits-busy-now.gsm' (language
'en')
-- Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/157-", &
> **
> Hi Michael,
>
> what if you reload the module chan_dahdi from within the * CLI? It should
> give some hints.
>
> Giorgio
>
>
>
> On 10/06/2011 05:22 PM, michael k wrote:
>
> Hi Giorgio,
>
> Thanks for your reply. I will produce so
at 8:43 PM, gincantalupo
wrote:
> **
> Hi Michael,
>
> if you type "dah" followed by TAB and nothing appears, it means you do not
> have dahdi module loaded or dahdi_cfg application not launched before
> starting asterisk.
>
> Giorgio
>
>
> On 10/06/2011 04:57 PM,
Hi All,
I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX
2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my
cli mode i am not getting the command *"dahdi show status"*
Output of CLI :
astrisks*CLI> *dahdi show status*
No such command 'dahdi show status
Hello All,
I have a pstn line can have the local, STD and ISD
capabilities. My local number is 91471-2527XXX and the region is India. I
would like to use the number for all possible calls ( local, STD and ISD
call facilities to Land line and mobile phones) through an FXO card
con
amportal.conf which tells FreePBX to convert ZAP into DAHDI.
>
>
>
> On Thu, Sep 29, 2011 at 11:57 AM, michael k wrote:
>
>> Can you please figure out the configuration issue in my freepbx ?
>>
>>
>>
>>
>>
>> On Thu, Sep 29, 2011 at 11:
any dial-able rule. See your FreePBX guide.
>
>
> On Thu, Sep 29, 2011 at 11:01 AM, michael k wrote:
>
>> Hi,
>>
>> Please see the sample.
>>
>> A ) Analog HardwareType Ports Action FXO Ports 1
>> Edit<http://192.168.1.134/admin/config.php?typ
ero on
'SIP/199-003a' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/199-003a'
On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind wrote:
> Some CLI logs will get you better help on the issue ! also paste
Hi All,
I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO card and installed in my asterisk server. My
freepbx detected the x100p FXO card and i can see the card specific details
in command line. I have configured the following things.
1. OUTBOUND c
Hi All,
I am new in asterisk. In my office we have purchased ISDN pri
line with 30 channels. we have more than 60 soft phone nodes and the
internal asterisk connectivity between extensions are working with soft
phones. Can anybody tell me which pci or pci express digium card can be use
Hi All,
Along with my asterisks server, all incoming calls to
my D-link DPH-80 ip phones are are working fine while calling from soft
phones with good voice clarity. But not able to make outgoing calls from the
same D-link DPH-80 ip phones to either soft phone or IP phone. What would be
t
Hi,
>
> Did you created your normal Inbound and Outbound routes in freepbx? For use
> with your zap channels?
>
> You'll problably have to change your routes on your pbx too...
>
> Regards,
>
> Carlos M Cruz
>
> 2011/7/28 michael k
>
>> Hello All,
&g
Hello All,
I don't even know the relevancy of my question. Please answer me if my
question have some sense.
I have recently implemented an asterisk server with freepbx. I have created
100 extentions and i can make successful calls between extensions from
anywhere. But my office have three differe
k/voicemail/default/199/unavail.tmp format: wav, 0x7917ad8
-- User cancelled message by pressing 0
-- Playing 'vm-sorry.ulaw' (language 'en')
-- Playing 'vm-torerecord.ulaw' (language 'en')
On Thu, Jun 30, 2011 at 9:42 PM, A J Stiles
.com] *On Behalf Of *michael k
> *Sent:* Thursday, June 30, 2011 10:09 AM
>
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Cannot figure out pound key in qwerty keyboard
>
>
> ** **
>
> All,
>
>I am new in Asterisk. I am using
digium.com] On Behalf Of
> > michael k
> > Sent: Thursday, June 30, 2011 11:23 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Cannot figure out pound key in
> > qwerty keyboard
> >
> > Hi All,
> >
> >
Hi All,
Thanks for the reply. I have typed Shift-3 (#) but the system keep on saying
that "I am sorry i did not understand your response". Any other solutions to
resolve this ?
On Thu, Jun 30, 2011 at 8:44 PM, Danny Nicholas wrote:
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
All,
I am new in Asterisk. I am using asterisks with freepbx 2.7.0.10
version. I have tried to setup voice mail by dialing *97 from my extension.
The prerecorded system asking for a "pond key" at the end of each recording.
But unfortunately i am not able to locate a pound key on my qwerty
It works.
I terminated the call during the playback.
AGI debug
AGI Tx >> 200 result=-1 endpos=480
HUP received!
Allowing
&setinuse() to get called
Thanks
Michael
On 10/6/05 12:13 AM, "Darren Wiebe" <[EMAIL PROTECTED]> wrote:
> Edit astcc.agi and stick these lines in before "sub load_conf
This is my debug with the same issue
The agi terminates during the "sub tell_time()"
and exits without calling "sub setinuse()" or completing the reset of the
script.
AGI Tx >> agi_request: astcc.agi
AGI Tx >> agi_channel: Zap/49-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> ag
I have been testing the ASTCC and have notice that when the caller hangs up
the line while the balance is being played back the sub savedata() is not
being called because the asterisk terminates the AGI and the rest of the
script does not get executed thus never returning:
AGI Script astcc.agi com
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-18 02:46:15 UTC
I am getting this error when the astcc.agi tries to UPDATE inuse = 0
LOG: unexpected EOF on client connection (postgres on Debian)
I use another astcc.agi that UPDATEs to a different server (Postgres
More info
On 8/17/05 3:34 AM, "Christoph Eicke" <[EMAIL PROTECTED]> wrote:
> Hi!
>
> I'm searching for a 1-800 number that simply plays music for a long time
> (>3mins) and no one picks up. I've bothered the AT&T lines so far when trying
> out my SIP->PSTN connection but then always someone ans
I have tested a fax call on asterisk with success. I used an IAXy on a
broadband Time Warner connection. Faxes are much more sensitive than voice
calls. If you have a good internet connection, faxes should complete fine.
The only downfall it is recommended that you call to verify fax transmission
a
I agree, why run to DBs. On the other hand, I have spoken with several
people asking about radius support for asterisk because they have a billing
solution that uses data from the radius servers to populate their billing
DB.
-Michael
On 3/17/05 11:00 AM, "Matthew Boehm" <[EMAIL PROTECTED]> wr
I had a similar problem with power.
I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up
and asterisk would load but the T1s on the Quad T1 card failed to come up. I
placed a loop on the card and still no change. Finally, I removed the UPS
and the T1s came up.
Do know if th
Does anyone know how does asterisk handles INFO digit from a PRI line?
Can info digit be used in extensions.conf to signal a call from a public
phone?
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If I am not mistaken, I believe the dial command is omitted if you do not
have a sound card configured on your system (loaded module).
-michael
On 12/2/04 1:07 AM, "Matt Hess" <[EMAIL PROTECTED]> wrote:
> Does cvs tag v1-0 not have a dial command? I do not seem to have one..
>> dial
> No such co
ECTED]
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Michael K. Rodriguez
Dialmex LLC
Director of Network Operations
200 S. 10th Suite 1209
McAllen, TX 78501
(956) 994-0014 x107 office
(95
Is it possible to send a call from the asterisk server to a
gateway via sipv2 protocol.
I have some 7960 phones that can receive a call from a
5350 via sipv2 and the phone can send to the gateway via sipv2.
Is there an exten that dials to a gateways ?
Michael
K
I am using a 7960 and it is registered to the *server, but I
keep getting this error. Does anyone know why?
NOTICE[5126]: File chan_sip.c, Line 3080 (handle_request):
Registration from 'sip:[EMAIL PROTECTED]' failed for '67.98.37.220'
Michael
K. Ro
NING[15374]: File app_record.c, Line 143 (record_exec):
Could not create file intro|gsm
extension.conf
exten => ,1,Record,intro|gsm
Thanks
Michael
K. Rodriguez
DialMex LLC
NOC Engineer
200 S. 10th Street
Suite 1209
McAllen, TX 78501
(956) 994-0014 x1
WARNING[21518]: File format_wav.c, Line 154 (check_header): Unexpected
freqency 22050
WARNING[21518]: File file.c, Line 346 (ast_filehelper): Unable to open
fd on intro
WARNING[21518]: File file.c, Line 553 (ast_streamfile): Unable to open
intro (format 12): No such file or directory
What does
:55 pm, Michael K. Rodriguez wrote:
> Any ideas on how to dialout exten => zap 1/1
Do you want to Dial the station at Zap/1? Or do you want to dial out
on the telephone line attached to Zap/1?
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[EMAIL PROTECTED
Any ideas on how to dialout exten => zap 1/1
Michael
K. Rodriguez
DialMex LLC
NOC Engineer
200 S. 10th Street
Suite 1209
McAllen, TX 78501
(956) 994-0014 x107
office
(956) 239-0627 mobile
(956) 682-5821 fax
[EMAIL PROTECTED]
Escalation
Procedure
+++
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