Hi to everybody,
I have a problem for received calls form my Grandstream HT-503.
I have a FXO connect to my PABX, and I can make a call from PABX to
VOIP, but I didn't received calls to my VOIP, to my PABX.
See the log:
Using SIP RTP CoS mark 5
-- Executing [27100@ramais:1]
Yes, UCARP is the problem about the sip ports
It conflict with ports IAX, SIP, RTP, etc.
Thanks man.
2016-08-29 21:44 GMT-03:00, Vitor Mazuco <vitor.maz...@gmail.com>:
> Humm right
>
> I think the UCARP can be the problem
>
> It is the problem about sip and rtp po
o:asterisk-users-
boun...@lists.digium.com] On Behalf Of Vitor Mazuco
Sent: Monday, August 29, 2016 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr
on reload
Hummm, but why It is with that problem?
I
sage is misleading
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Mazuco
> Sent: Monday, August 29, 2016 10:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subjec
Sorry,
I just see warning.
2016-08-29 11:40 GMT-03:00, Vitor Mazuco <vitor.maz...@gmail.com>:
> I just see warning?
>
>
> 2016-08-29 11:30 GMT-03:00, Telium Technical Support <supp...@telium.ca>:
>> This shows that asterisk's IAX is already bound to all a
l Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Mazuco
> Sent: Monday, August 29, 2016 8:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX UNRE
Hi, see the log below
root@AsteriskSlave:~# ip addr
1: lo: mtu 65536 qdisc noqueue state UNKNOWN
group default
link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
inet 127.0.0.1/8 scope host lo
valid_lft forever preferred_lft forever
inet6 ::1/128 scope
Hi, I have already tried to change for bindaddr=0.0.0.0
but it didn't worked.
2016-08-26 11:44 GMT-03:00, Frank Vanoni <mailingl...@linuxista.com>:
> On Fri, 2016-08-26 at 10:12 -0300, Vitor Mazuco wrote:
>
>> bindaddr = all
>
> T
Hi to everybody,
My IAX is not working, When I type reload IAX it returns me:
AsteriskSlave*CLI> iax2 reload
== Parsing '/etc/asterisk/iax.conf': Found
== Parsing '/etc/asterisk/users.conf': Found
[Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13546 set_config:
Ignoring bindport on reload
[Aug
So, PlaySMS do not working with ChanDongle?
2016-07-15 12:34 GMT-03:00, Emiliano Vazquez <emilianovazq...@gmail.com>:
> El 15/07/16 a las 12:00, Vitor Mazuco escribió:
>> Hi!
>>
>> I have a chan dongle and I want to use PlaySMS with Chan Dongle for
>> send many
Hi!
I have a chan dongle and I want to use PlaySMS with Chan Dongle for
send many SMS per day.
Is possible to use this?
Thanks
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Have I to fallow this tutorial?
http://www.ruchirablog.com/unlock-voice-huawei-hspa/
2016-06-01 18:30 GMT-03:00, Vitor Mazuco <vitor.maz...@gmail.com>:
> Hi to everybody
>
> I have a Huawei E160E but it not works in my chan dongle, see the log
>
> == Using SIP RTP CoS mark
Hi to everybody
I have a Huawei E160E but it not works in my chan dongle, see the log
== Using SIP RTP CoS mark 5
-- Executing [951729377@ramais:1] Dial("SIP/2002-",
"Dongle/dongle0/951729377,60,tT") in new stack
-- Called Dongle/dongle0/9
[Jun 1 18:24:40] ERROR[5707]:
humm, ok.
Thanks very much
2016-05-27 19:56 GMT-03:00, Richard Mudgett <rmudg...@digium.com>:
> On Fri, May 27, 2016 at 5:28 PM, Vitor Mazuco <vitor.maz...@gmail.com>
> wrote:
>
>> Hi to everybody
>>
>> my system is be attack, but I dont know what this
Hi to everybody
my system is be attack, but I dont know what this means
[May 27 15:12:24] WARNING[26018] chan_skinny.c: Partial data received,
waiting (76 bytes read of 786)
[chan_skinny.c] skinny_session[0][C-] skinny_session:
WARNING[May 27 15:52:32] Asterisk 13.8.0 built by root @
Hello everyone
I have a TDM 800 on an Ubuntu Server
he's just getting call normally, but when I call any number by this
board, it is silent and not make the call.
look at the log
Executing [629886874@ramais:1] Dial("SIP/2000-000e",
"DAHDI/6-1/29xxx,60,tT") in new stack
[May 18
see the site here https://www.voipraider.com/calling_rates/
2016-05-09 19:43 GMT-03:00, Vitor Mazuco <vitor.maz...@gmail.com>:
> VoipRaider the site, says calls to landlines in Brazil is FREE within
> the freedays period. Log in to the website and hire the service, it
> says that
VoipRaider the site, says calls to landlines in Brazil is FREE within
the freedays period. Log in to the website and hire the service, it
says that I have 90 days of freedays paying for cheaper service is $
10.. That is from what I understand, I will pay 10 dolares for
unlimited call in landlines
Hello to everyone
I have a Automatic Call Distribution for I receive calls, and it is normal
But how can I make for outbound calls using a E1 links with 30 channels?
Is there a specific code for that?
Thanks
--
_
--
Humm thanks for your reply,
Do you know whats is step for I can transform this card link a fax modem?
2016-03-30 9:36 GMT-03:00, A J Stiles <asterisk_l...@earthshod.co.uk>:
> On Wednesday 30 Mar 2016, Vitor Mazuco wrote:
>> Hi!
>>
>> Is possible to use X100p TDM4
Hi!
Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or
any others digium card FXO for use Fax modem?
Thanks.
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Is possible with Telegram?
2016-03-29 9:39 GMT-03:00, Emiliano Vazquez :
> El 29/03/16 a las 08:29, Steve Howes escribió:
>> I don't think you can. Whatsapp is a closed system.
>>
>> Steve
> And they change your code every day and make it always obfuscated.
>
>
Is possible to use Asterisk or Chan Dongle like this topology of what
we do, basically a RAS server that receives call from mobile terminals
data, closes a PPP and offers these terminals the possibility of
access to an IP network.
Lile this pic
From: asterisk-users-boun...@lists.digium.com <
asterisk-users-boun...@lists.digium.com> on behalf of Vitor Mazuco <
vitor.maz...@gmail.com>
Sent: Wednesday, March 9, 2016 19:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Ho
Hi !
I want to install a Telephone IP Avaya 4610SW in my Asterisk 11, but I
cant install.
It asks a TFTP/HTTP Server, but is necessary I install it in mu
Asterisk Server for works my Telephone?
The manual is here https://downloads.avaya.com/css/P8/documents/003880182
Thanks in advanced.
--
sthan - 302031 , India*
> *Mobile No: +917597056895 <%2B917597056895>*
>
> On Sat, Mar 5, 2016 at 2:15 AM, Vitor Mazuco <vitor.maz...@gmail.com>
> wrote:
>
>> Hi!
>>
>> How can I setup my Chan Dongle recived calls in my Asterisk?
>>
>> I have
Hi!
How can I setup my Chan Dongle recived calls in my Asterisk?
I have to setup in dongle.conf ? Or in extensions.conf?
And the code for recive I found this site
http://asterisk-service.com/page/chan-dongle-use
I have to To save Subscriber Number before?
See the error log in my Asterisk
Humm ok
But my monden not appear in /dev/ and it not show like ttyUSB
I have to install the driver before? Or is not necessary?
Thanks in advanced
Em 03/03/2016 06:13, "Frank Vanoni" <mailingl...@linuxista.com> escreveu:
> On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco w
Hi everyone!
I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but
my Huawei E153 is not working in my Asterisk.
I fallow this rules
http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14
But not successes.
Thanks in advanced,
--
Hello everyone, I have some problems to enable push the Zoiper
Windows Phone in my Asterisk 11.
Below is the result of CLI
== Using SIP RTP CoS mark 5
-- Executing [1033@ramais:1] Answer("SIP/1030-0201", "") in new stack
> 0x7efc90024190 -- Probation passed - setting RTP source
Right,
I'll use in Asterisk 11 and I reply for you.
Thanks,
2016-02-12 14:35 GMT-02:00, Shabbir abbasi <shabbirabbas...@gmail.com>:
> i have not tested asterik 13
> but try this
> core set debug 10
> and look what is hapening
>
> On Fri, Feb 12, 2016 at 9:33 PM,
Hi!
I'm trying to use dongle in my Asterisk
But appear for me all time this error
[Feb 12 13:49:10] ERROR[13347]: chan_dongle.c:442 do_monitor_phone:
[dongle0] timedout while waiting 'OK' in response to 'AT'
-- [dongle0] Error initializing Dongle
-- [dongle0] Dongle has disconnected
Yes, I used IMEI.
But in CLI appearing nothing and it not register.
2016-02-12 14:27 GMT-02:00, Shabbir abbasi <shabbirabbas...@gmail.com>:
> have you tried imei discovery
> imei=123456789012345
>
>
> write imei number instaed of 12345...
>
> On Fri, Feb 12, 20
imei=123456789012345
>
> and imei exact same as on your device ?
>
> On Fri, Feb 12, 2016 at 9:29 PM, Vitor Mazuco <vitor.maz...@gmail.com>
> wrote:
>
>> Yes, I used IMEI.
>>
>> But in CLI appearing nothing and it not register.
>>
>>
>>
&
and imsi must contain exactly 15 digits !
; imei/imsi discovery is available on Linux only
imei=352098043831724
;imsi=123456789012345
My imei is 352098043831724
But nothing change.
2016-02-12 15:12 GMT-02:00, Frank <mailingl...@linuxista.com>:
> On Fri, 2016-02-12 at 14:33 -0200, Vit
I think that my monden is locked for Voice
I use a Huawei E173, someone know how can I unlock it?
Is necessary to upgrade the firmware?
2016-02-12 15:39 GMT-02:00, Vitor Mazuco <vitor.maz...@gmail.com>:
> I tried this
>
> [dongle0]
> ;audio=/dev/ttyUSB1 ;
Hi everybody!
Is possible to integrate WhatsApp VoIP on Asterisk?
Or is there some tricks for that? Like Yowsup?
Thanks in advanced.
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Humm thanks very much :)
Em 04/02/2016 19:58, "Doug Lytle" <supp...@drdos.info> escreveu:
>
>
> >>> On Feb 4, 2016, at 12:55 PM, Vitor Mazuco vitor.maz...@gmail.com
> wrote:
>
> >>> so this context parkedcal
Humm,
so this context parkedcalls is inside on features.conf?
2016-02-03 17:42 GMT-02:00, Doug Lytle <supp...@drdos.info>:
>>>> On Feb 3, 2016, at 2:32 PM, Vitor Mazuco vitor.maz...@gmail.com wrote:
>
>>>> Ah no, I'm asking what code I put inside of parkedca
17:27 GMT-02:00, Doug Lytle <supp...@drdos.info>:
>>>> On Feb 3, 2016, at 2:19 PM, Vitor Mazuco vitor.maz...@gmail.com wrote:
>
>>>> Humm, thanks for your reply
>>>> But whats is the code in parkedcalls context.
>>>> Please, can y
Hi!
I tried to use Parking Calls
I use Asterisk 13, but I can't park any calls and it returns me
[Feb 3 16:56:11] WARNING[1693]: pbx.c:12543
ast_context_verify_includes: Context 'ramais' tries to include
nonexistent context 'parkedcalls'
What is the correct code for put in extensions.conf?
Humm, thanks for your reply
But whats is the code in parkedcalls context.
Please, can you give an example?
Thanks very much.
2016-02-03 17:15 GMT-02:00, Richard Mudgett <rmudg...@digium.com>:
> On Wed, Feb 3, 2016 at 1:05 PM, Vitor Mazuco <vitor.maz...@gmail.com>
> wrote
Hi everybody!
I'm trying to install a CDR Viewer https://github.com/g613/asterisk-cdr-viewer
I'ts work well, but my problem is to make Asterisk store to MySQL using ODBC
When I make a call the CLI returns for me
See the log:
== Using SIP RTP CoS mark 5
-- Executing [2021@ramais:1]
Hi everybody,
My Asterisk, all time appear this log
[Jan 7 15:37:04] ERROR[1174] chan_sip.c: The To header was truncated
in call '6c66e5b6058ae257003c0f7e778da0fe@191.x'. This call
setup will fail.
[Jan 7 15:37:18] ERROR[1174] chan_sip.c: The To header was truncated
in call
Hi everyone!
I have a Digium Card TDM410
But, it appear for me this massege
chan_dahdi.c:8061 handle_alarms: Detected alarm on channel 3: Red Alarm
But my line is ok!
But sometimes it back
sig_analog.c:3807 analog_handle_init_event: Alarm cleared on channel 2
But it again back to red alarm.
My line is comming from a monden ADSL that it provide internet too.
2016-01-05 12:46 GMT-02:00, Ryan, Travis <ry...@oscarwinski.com>:
>
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.
; > > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > > boun...@lists.digium.com] On Behalf Of Vitor Mazuco
> > > Sent: Tuesday, January 05, 2016 9:21 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subjec
Humm, if I put a filter in this lines, maybe back?
2016-01-05 12:36 GMT-02:00, Ryan, Travis <ry...@oscarwinski.com>:
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Vi
I use, Ubuntu Server
2015-12-23 11:31 GMT-02:00, er ic :
> What is the best asterisk platform to use? What are you guys using?
>
> I am looking for something to host either in our data center or at the
> customer prem where I have the control over the unit and not
lain, not
> to mention for sizing.
>
>
>
> On 12/16/15 8:23 AM, Vitor Mazuco wrote:
>>
>> Humm whats is the diferent?
>>
>> Em 16/12/2015 14:19, "Annus Fictus" <annusfic...@gmail.com
>> <mailto:annusfic...@gmail.com>> escreveu:
>>
>
Hi everyone!
I'm trying to install a database using the asterisk-CDR-viewer. It
uses MySQL and I'm using Asterisk 11.I know that it needs to
synchronize with the ODBC database.
But I'm in trouble, it shows an error message will play when the
database "cdr_odbc.c: 163 odbc_log:. Unable to
Hi everyone!
I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult.
Is there others optins for billing?
Thanks
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Humm whats is the diferent?
Em 16/12/2015 14:19, "Annus Fictus" <annusfic...@gmail.com> escreveu:
> CDR-STATS is for reporting.
>
> A2Billing is for billing...
>
> Regards
>
> El 16/12/2015 a las 11:15, Vitor Mazuco escribió:
>
>> Hi everyone!
>&g
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