[asterisk-users] Received response: "Forbidden" in Grandstream HT-503

[Vitor Mazuco]

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] MixMonitor("SIP/2000-bd8b",
"/media/HDExterno/gravacoes/feitas/APTO/27100/1489675579.120546.wav")
in new stack
-- Executing [27100@ramais:2] Dial("SIP/2000-bd8b",
"SIP/136/100,60,tT") in new stack
  == Begin MixMonitor Recording SIP/2000-bd8b
  == Using SIP RTP CoS mark 5
-- Called SIP/136/100
[2017-03-16 11:46:19] WARNING[1554][C-98b9]: chan_sip.c:23843
handle_response_invite: Received response: "Forbidden" from
';tag=as57804b2e'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2000-bd8b' status is 'CHANUNAVAIL'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/2000-bd8b


And the SIP Debuug:

Called SIP/136/100

<--- SIP read from UDP:192.168.25.169:3329 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089
From: ;tag=as62bede9e
To: 
Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.25.169:3329 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089
From: ;tag=as62bede9e
To: ;tag=1820807938
Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.25.169:3329:
ACK sip:100@192.168.25.169 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport
Max-Forwards: 70
From: ;tag=as62bede9e
To: ;tag=1820807938
Contact: 
Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.10.0
Content-Length: 0


---
[2017-03-16 11:34:53] WARNING[1554][C-98af]: chan_sip.c:23843
handle_response_invite: Received response: "Forbidden" from
';tag=as62bede9e'
Scheduling destruction of SIP dialog
'692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089' in 6400 ms
(Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2000-bd7a' status is 'CHANUNAVAIL'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/2000-bd7a

See my sip.conf

;;
[136]
type=friend
defaultuser=136
secret=X
qualify=yes
;nat=no
nat=force_rport,comedia
context=ramais
;insecure=invite,port
disallow=all
allow=ulaw,alaw,gsm
host=dynamic
canreinvite=no
regext=136
callgroup=1
pickupgroup=1

I have a LOAD BALANCE too in this Grandstream.

The problem is the NAT/Firewall? Because the FXS is working well.

Thanks in advanced!

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Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

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 :
> Humm right
>
> I think the UCARP can be the problem
>
> It is the problem about sip and rtp ports
>
> I will remove it and make the tests
>
> Thanks man
>
> Em 29/08/2016 14:19, "Telium Technical Support" 
> escreveu:
>
> Possibly - I noticed this thread only in the context of an IAX problem.  I
> can't speak to UCARP
>
> If you're trying to my a high availability cluster out of Asterisk servers
> have a look at http://serverfault.com/questions/733403/high-
> availability-asterisk-options/733441#733441
>
>
> -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 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 use UCARP, maybe is this the problem?
>
>
> 2016-08-29 12:17 GMT-03:00, Telium Technical Support :
>> Oh!  In that case ignore it.
>>
>> Asterisk won't rebind the adapter if you've only changed parameters.  The
>> message 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
>> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring
>> bindport/bindaddr
>> on reload
>>
>> Sorry,
>>
>> I just see warning.
>>
>>
>>
>> 2016-08-29 11:40 GMT-03:00, Vitor Mazuco :
>>> I just see  warning?
>>>
>>>
>>> 2016-08-29 11:30 GMT-03:00, Telium Technical Support
>>> :
>>>> This shows that asterisk's IAX is already bound to all adapters - so
>>>> that's
>>>> good.  Symptomatically does your IAX stop working?  Or do you just see
>>>> a
>>>> warning?
>>>>
>>>> -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 8:46 AM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring
>>>> bindport/bindaddr
>>>> on reload
>>>>
>>>> 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 host
>>>>valid_lft forever preferred_lft forever
>>>> 2: p3p1:  mtu 1500 qdisc noop state DOWN group
>>>> default qlen 1000
>>>> link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
>>>> 3: p4p1:  mtu 1500 qdisc pfifo_fast
>>>> state UP group default qlen 1000
>>>> link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
>>>> inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
>>>>valid_lft forever preferred_lft forever
>>>> inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
>>>>valid_lft 86398sec preferred_lft 43198sec
>>>> inet6 fe80::2e0:4cff:fe44:195/64 scope link
>>>>valid_lft forever preferred_lft forever
>>>> 4: p5p1:  mtu 1500 qdisc noop state DOWN group
>>>> default qlen 1000
>>>> link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff
>>>>
>>>> and
>>>>
>>>> root@AsteriskSlave:~# netstat -anp | grep ast
>>>> tcp0  0 0.0.0.0:20000.0.0.0:*
>>>> OUÇA   2050/asterisk
>>>> tcp0  0 0.0.0.0:53380.0.0.0:*
>>>> OUÇA   2050/asterisk
>>>> udp0  0 0.0.0.0:38180   0.0.0.0:*
>>>>  2050/asterisk
>>>> udp0  0 0.0.0.0:45200.0.0.0:*
>>>>  2050/asterisk
>>>> udp0  0 0.0.0.0:4659

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

Humm right

I think the UCARP can be the problem

It is the problem about sip and rtp ports

I will remove it and make the tests

Thanks man

Em 29/08/2016 14:19, "Telium Technical Support" 
escreveu:

Possibly - I noticed this thread only in the context of an IAX problem.  I
can't speak to UCARP

If you're trying to my a high availability cluster out of Asterisk servers
have a look at http://serverfault.com/questions/733403/high-
availability-asterisk-options/733441#733441


-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 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 use UCARP, maybe is this the problem?


2016-08-29 12:17 GMT-03:00, Telium Technical Support :
> Oh!  In that case ignore it.
>
> Asterisk won't rebind the adapter if you've only changed parameters.  The
> message 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
> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr
> on reload
>
> Sorry,
>
> I just see warning.
>
>
>
> 2016-08-29 11:40 GMT-03:00, Vitor Mazuco :
>> I just see  warning?
>>
>>
>> 2016-08-29 11:30 GMT-03:00, Telium Technical Support :
>>> This shows that asterisk's IAX is already bound to all adapters - so
>>> that's
>>> good.  Symptomatically does your IAX stop working?  Or do you just see a
>>> warning?
>>>
>>> -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 8:46 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring
>>> bindport/bindaddr
>>> on reload
>>>
>>> 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 host
>>>valid_lft forever preferred_lft forever
>>> 2: p3p1:  mtu 1500 qdisc noop state DOWN group
>>> default qlen 1000
>>> link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
>>> 3: p4p1:  mtu 1500 qdisc pfifo_fast
>>> state UP group default qlen 1000
>>> link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
>>> inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
>>>valid_lft forever preferred_lft forever
>>> inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
>>>valid_lft 86398sec preferred_lft 43198sec
>>> inet6 fe80::2e0:4cff:fe44:195/64 scope link
>>>valid_lft forever preferred_lft forever
>>> 4: p5p1:  mtu 1500 qdisc noop state DOWN group
>>> default qlen 1000
>>> link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff
>>>
>>> and
>>>
>>> root@AsteriskSlave:~# netstat -anp | grep ast
>>> tcp0  0 0.0.0.0:20000.0.0.0:*
>>> OUÇA   2050/asterisk
>>> tcp0  0 0.0.0.0:53380.0.0.0:*
>>> OUÇA   2050/asterisk
>>> udp0  0 0.0.0.0:38180   0.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:45200.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:46590.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:27270.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:50000.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:50890.0.0.0:*
>>>  2050/asterisk
>>> unix  2  [ ACC ] STREAM OUVINDO   484
>>> 2050/asterisk   /var/run/asterisk/asterisk.ctl
>>> unix  2  [ ] DGRAM116862050/asterisk
>>>
>>> 2016-08-26 19:21 GMT-03:00, Telium Technical Support :
>>

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

Hummm, but why It is with that problem?

I use UCARP, maybe is this the problem?


2016-08-29 12:17 GMT-03:00, Telium Technical Support :
> Oh!  In that case ignore it.
>
> Asterisk won't rebind the adapter if you've only changed parameters.  The
> message 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
> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr
> on reload
>
> Sorry,
>
> I just see warning.
>
>
>
> 2016-08-29 11:40 GMT-03:00, Vitor Mazuco :
>> I just see  warning?
>>
>>
>> 2016-08-29 11:30 GMT-03:00, Telium Technical Support :
>>> This shows that asterisk's IAX is already bound to all adapters - so
>>> that's
>>> good.  Symptomatically does your IAX stop working?  Or do you just see a
>>> warning?
>>>
>>> -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 8:46 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring
>>> bindport/bindaddr
>>> on reload
>>>
>>> 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 host
>>>valid_lft forever preferred_lft forever
>>> 2: p3p1:  mtu 1500 qdisc noop state DOWN group
>>> default qlen 1000
>>> link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
>>> 3: p4p1:  mtu 1500 qdisc pfifo_fast
>>> state UP group default qlen 1000
>>> link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
>>> inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
>>>valid_lft forever preferred_lft forever
>>> inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
>>>valid_lft 86398sec preferred_lft 43198sec
>>> inet6 fe80::2e0:4cff:fe44:195/64 scope link
>>>valid_lft forever preferred_lft forever
>>> 4: p5p1:  mtu 1500 qdisc noop state DOWN group
>>> default qlen 1000
>>> link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff
>>>
>>> and
>>>
>>> root@AsteriskSlave:~# netstat -anp | grep ast
>>> tcp0  0 0.0.0.0:20000.0.0.0:*
>>> OUÇA   2050/asterisk
>>> tcp0  0 0.0.0.0:53380.0.0.0:*
>>> OUÇA   2050/asterisk
>>> udp0  0 0.0.0.0:38180   0.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:45200.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:46590.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:27270.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:50000.0.0.0:*
>>>  2050/asterisk
>>> udp0  0 0.0.0.0:50890.0.0.0:*
>>>  2050/asterisk
>>> unix  2  [ ACC ] STREAM OUVINDO   484
>>> 2050/asterisk   /var/run/asterisk/asterisk.ctl
>>> unix  2  [ ] DGRAM116862050/asterisk
>>>
>>> 2016-08-26 19:21 GMT-03:00, Telium Technical Support :
>>>> Could you post the result of "ip addr" command, and "netstat -anp | grep
>>>> ast" after the reload?
>>>>
>>>> I suspect something else is going on here...
>>>>
>>>>
>>>>
>>>> --
>>>> _
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>>>   http://www.asterisk.org/community/astricon-user-conference
>>>>
>>>> New to Asterisk? Start here:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
>>>> aste

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

Sorry,

I just see warning.



2016-08-29 11:40 GMT-03:00, Vitor Mazuco :
> I just see  warning?
>
>
> 2016-08-29 11:30 GMT-03:00, Telium Technical Support :
>> This shows that asterisk's IAX is already bound to all adapters - so
>> that's
>> good.  Symptomatically does your IAX stop working?  Or do you just see a
>> warning?
>>
>> -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 8:46 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring
>> bindport/bindaddr
>> on reload
>>
>> 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 host
>>valid_lft forever preferred_lft forever
>> 2: p3p1:  mtu 1500 qdisc noop state DOWN group
>> default qlen 1000
>> link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
>> 3: p4p1:  mtu 1500 qdisc pfifo_fast
>> state UP group default qlen 1000
>> link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
>> inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
>>valid_lft forever preferred_lft forever
>> inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
>>valid_lft 86398sec preferred_lft 43198sec
>> inet6 fe80::2e0:4cff:fe44:195/64 scope link
>>valid_lft forever preferred_lft forever
>> 4: p5p1:  mtu 1500 qdisc noop state DOWN group
>> default qlen 1000
>> link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff
>>
>> and
>>
>> root@AsteriskSlave:~# netstat -anp | grep ast
>> tcp0  0 0.0.0.0:20000.0.0.0:*
>> OUÇA   2050/asterisk
>> tcp0  0 0.0.0.0:53380.0.0.0:*
>> OUÇA   2050/asterisk
>> udp0  0 0.0.0.0:38180   0.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:45200.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:46590.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:27270.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:50000.0.0.0:*
>>  2050/asterisk
>> udp0  0 0.0.0.0:50890.0.0.0:*
>>  2050/asterisk
>> unix  2  [ ACC ] STREAM OUVINDO   484
>> 2050/asterisk   /var/run/asterisk/asterisk.ctl
>> unix  2  [ ] DGRAM116862050/asterisk
>>
>> 2016-08-26 19:21 GMT-03:00, Telium Technical Support :
>>> Could you post the result of "ip addr" command, and "netstat -anp | grep
>>> ast" after the reload?
>>>
>>> I suspect something else is going on here...
>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>>   http://www.asterisk.org/community/astricon-user-conference
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>   http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriC

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

I just see  warning?


2016-08-29 11:30 GMT-03:00, Telium Technical Support :
> This shows that asterisk's IAX is already bound to all adapters - so that's
> good.  Symptomatically does your IAX stop working?  Or do you just see a
> warning?
>
> -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 8:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr
> on reload
>
> 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 host
>valid_lft forever preferred_lft forever
> 2: p3p1:  mtu 1500 qdisc noop state DOWN group
> default qlen 1000
> link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
> 3: p4p1:  mtu 1500 qdisc pfifo_fast
> state UP group default qlen 1000
> link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
> inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
>valid_lft forever preferred_lft forever
> inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
>valid_lft 86398sec preferred_lft 43198sec
> inet6 fe80::2e0:4cff:fe44:195/64 scope link
>valid_lft forever preferred_lft forever
> 4: p5p1:  mtu 1500 qdisc noop state DOWN group
> default qlen 1000
> link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff
>
> and
>
> root@AsteriskSlave:~# netstat -anp | grep ast
> tcp0  0 0.0.0.0:20000.0.0.0:*
> OUÇA   2050/asterisk
> tcp0  0 0.0.0.0:53380.0.0.0:*
> OUÇA   2050/asterisk
> udp0  0 0.0.0.0:38180   0.0.0.0:*
>  2050/asterisk
> udp0  0 0.0.0.0:45200.0.0.0:*
>  2050/asterisk
> udp0  0 0.0.0.0:46590.0.0.0:*
>  2050/asterisk
> udp0  0 0.0.0.0:27270.0.0.0:*
>  2050/asterisk
> udp0  0 0.0.0.0:50000.0.0.0:*
>  2050/asterisk
> udp0  0 0.0.0.0:50890.0.0.0:*
>  2050/asterisk
> unix  2  [ ACC ] STREAM OUVINDO   484
> 2050/asterisk   /var/run/asterisk/asterisk.ctl
> unix  2  [ ] DGRAM116862050/asterisk
>
> 2016-08-26 19:21 GMT-03:00, Telium Technical Support :
>> Could you post the result of "ip addr" command, and "netstat -anp | grep
>> ast" after the reload?
>>
>> I suspect something else is going on here...
>>
>>
>>
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Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

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 host
   valid_lft forever preferred_lft forever
2: p3p1:  mtu 1500 qdisc noop state DOWN group
default qlen 1000
link/ether 00:13:3b:12:02:21 brd ff:ff:ff:ff:ff:ff
3: p4p1:  mtu 1500 qdisc pfifo_fast
state UP group default qlen 1000
link/ether 00:e0:4c:44:01:95 brd ff:ff:ff:ff:ff:ff
inet 192.168.25.25/24 brd 192.168.25.255 scope global p4p1
   valid_lft forever preferred_lft forever
inet6 2804:7f1:4080:fe45:2e0:4cff:fe44:195/64 scope global dynamic
   valid_lft 86398sec preferred_lft 43198sec
inet6 fe80::2e0:4cff:fe44:195/64 scope link
   valid_lft forever preferred_lft forever
4: p5p1:  mtu 1500 qdisc noop state DOWN group
default qlen 1000
link/ether 14:dd:a9:82:38:ff brd ff:ff:ff:ff:ff:ff

and

root@AsteriskSlave:~# netstat -anp | grep ast
tcp0  0 0.0.0.0:20000.0.0.0:*
OUÇA   2050/asterisk
tcp0  0 0.0.0.0:53380.0.0.0:*
OUÇA   2050/asterisk
udp0  0 0.0.0.0:38180   0.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:45200.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:46590.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:27270.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:50000.0.0.0:*
 2050/asterisk
udp0  0 0.0.0.0:50890.0.0.0:*
 2050/asterisk
unix  2  [ ACC ] STREAM OUVINDO   484
2050/asterisk   /var/run/asterisk/asterisk.ctl
unix  2  [ ] DGRAM116862050/asterisk

2016-08-26 19:21 GMT-03:00, Telium Technical Support :
> Could you post the result of "ip addr" command, and "netstat -anp | grep
> ast" after the reload?
>
> I suspect something else is going on here...
>
>
>
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Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

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 :
> On Fri, 2016-08-26 at 10:12 -0300, Vitor Mazuco wrote:
>
>> bindaddr = all
>
> Try:
>
> bindaddr=0.0.0.0
>
>
>
>
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[asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

[Vitor Mazuco]

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 26 10:05:04] NOTICE[18078]: chan_iax2.c:13610 set_config:
Ignoring bindaddr on reload

And the peers is not working:

Name/UsernameHost   Mask
   Port   Status
Description
prote1-prote2/p  192.168.25.26(S)
255.255.255.255   4569  (T)  UNREACHABLE
1 iax2 peers [0 online, 1 offline, 0 unmonitored]


See my both iax.conf

SERVER 1
[General]
bindport=4659
bindaddr = all
disallow=all
allow=ulaw;alaw
;Contas para os servidores das filiais.
;;
[prote1-prote2]
secret= password
username=prote1-prote2
host=192.168.25.26
type=friend
context=ramais
qualify=yes
trunk=yes
auth = md5
;;


SERVER 2

[General]
bindport=4659
bindaddr=all
disallow=all
allow=ulaw;alaw
;Contas para os servidores das filiais.
;;
[prote1-prote2]
secret= password
username=prote1-prote2
host=192.168.25.25
type=friend
context=ramais
qualify=yes
trunk=yes
auth = md5
;;
;;


How can I fix it?


Thanks.

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Re: [asterisk-users] PlaySMS with Chan Dongle?

[Vitor Mazuco]

So, PlaySMS do not working with ChanDongle?



2016-07-15 12:34 GMT-03:00, Emiliano Vazquez :
> 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 SMS per day.
>>
>>
>> Is possible to use this?
>>
>>
>> Thanks
> You can't share the same dev two times. I will have conflicts some day.
> I don't know how playSMS send at commands but i think i will use in
> exclusive way like chan_dongle.
>
> You can write your owns scripts to send over Chan_dongle to send and
> receive or make a try to use PlaySMS and connect to modem over chan_dongle.
>
> Best regards.
>
> Emiliano.
>
>
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[asterisk-users] PlaySMS with Chan Dongle?

[Vitor Mazuco]

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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Re: [asterisk-users] Chan Dongle AT^DDSETEX failed

[Vitor Mazuco]

Have I to fallow this tutorial?

http://www.ruchirablog.com/unlock-voice-huawei-hspa/



2016-06-01 18:30 GMT-03:00, Vitor Mazuco :
> 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]: at_response.c:467 at_response_error:
> [dongle0] Dial failed
> -- Dongle/dongle0-01 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
> -- Auto fallthrough, channel 'SIP/2002-' status is 'CONGESTION'
> [Jun  1 18:24:40] ERROR[5707]: at_response.c:472 at_response_error:
> [dongle0] AT^DDSETEX failed
>
>
> It is problem of unlock my mondem or it is the problem of my Asterisk?
>

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[asterisk-users] Chan Dongle AT^DDSETEX failed

[Vitor Mazuco]

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]: at_response.c:467 at_response_error:
[dongle0] Dial failed
-- Dongle/dongle0-01 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/2002-' status is 'CONGESTION'
[Jun  1 18:24:40] ERROR[5707]: at_response.c:472 at_response_error:
[dongle0] AT^DDSETEX failed


It is problem of unlock my mondem or it is the problem of my Asterisk?

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Re: [asterisk-users] What this attacks means?

[Vitor Mazuco]

humm, ok.

Thanks very much

2016-05-27 19:56 GMT-03:00, Richard Mudgett :
> On Fri, May 27, 2016 at 5:28 PM, Vitor Mazuco 
> wrote:
>
>> Hi to everybody
>>
>> my system is be attack, but I dont know what this means
>>
>
> 
>
>>
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
>> 'nat' for a peer/user that differs from the  global setting can make
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
>> peer/user discoverable by an attacker. Replies for non-existent
>> peers/users
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
>> different port than replies for an existing peer/user. If at all
>> possible,
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
>> setting and do not set 'nat' per peer/user.
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='132'
>> global force_rport='No' peer/user force_rport='Yes')
>>
>
>
>
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
>> 'nat' for a peer/user that differs from the  global setting can make
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
>> peer/user discoverable by an attacker. Replies for non-existent
>> peers/users
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
>> different port than replies for an existing peer/user. If at all
>> possible,
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
>> setting and do not set 'nat' per peer/user.
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='133'
>> global force_rport='No' peer/user force_rport='Yes')
>>
>
>
>
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
>> 'nat' for a peer/user that differs from the  global setting can make
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
>> peer/user discoverable by an attacker. Replies for non-existent
>> peers/users
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
>> different port than replies for an existing peer/user. If at all
>> possible,
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
>> setting and do not set 'nat' per peer/user.
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='134'
>> global force_rport='No' peer/user force_rport='Yes')
>>
>
>
>
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
>> 'nat' for a peer/user that differs from the  global setting can make
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
>> peer/user discoverable by an attacker. Replies for non-existent
>> peers/users
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
>> different port than replies for an existing peer/user. If at all
>> possible,
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
>> setting and do not set 'nat' per peer/user.
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='135'
>> global force_rport='No' peer/user force_rport='Yes')
>>
>
>
>
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
>> 'nat' for a peer/user that differs from the  global setting can make
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
>> peer/user discoverable by an attacker. Replies for non-existent
>> peers/users
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
>> different port than replies for an existing peer/user. If at all
>> possible,
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
>> setting and do not set 'nat' per peer/user.
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='136'
>> global force_rport='No' peer/user force_rport='Yes')
>>
>
>
>
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
>> 'nat' for a peer/user that differs from the  global setting can make
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
>> peer/user discoverable by an attacker. Replies for non-existent
>> peers/users
>> [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
>> different port than replies for an existing peer/user. 

[asterisk-users] What this attacks means?

[Vitor Mazuco]

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 @ asterisk on a
x86_64 running Linux on 2016-04-04 19:02:51 UTC
[May 27 15:52:32] NOTICE[2306] cdr.c: CDR simple logging enabled.
[May 27 15:52:32] NOTICE[2306] loader.c: 234 modules will be loaded.
[May 27 15:52:32] WARNING[2306] res_phoneprov.c: Unable to find a
valid server address or name.
[May 27 15:52:32] ERROR[2306] ari/config.c: No configured users for ARI
[May 27 15:52:33] NOTICE[2306] chan_skinny.c: Configuring skinny from
skinny.conf
[May 27 15:52:33] WARNING[2306] chan_dahdi.c: Ignoring any changes to
'userbase' (on reload) at line 23.
[May 27 15:52:33] WARNING[2306] chan_dahdi.c: Ignoring any changes to
'vmsecret' (on reload) at line 31.
[May 27 15:52:33] WARNING[2306] chan_dahdi.c: Ignoring any changes to
'hassip' (on reload) at line 35.
[May 27 15:52:33] WARNING[2306] chan_dahdi.c: Ignoring any changes to
'hasiax' (on reload) at line 39.
[May 27 15:52:33] WARNING[2306] chan_dahdi.c: Ignoring any changes to
'hasmanager' (on reload) at line 47.
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
'nat' for a peer/user that differs from the  global setting can make
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
peer/user discoverable by an attacker. Replies for non-existent
peers/users
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all
possible,
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='132'
global force_rport='No' peer/user force_rport='Yes')
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
'nat' for a peer/user that differs from the  global setting can make
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
peer/user discoverable by an attacker. Replies for non-existent
peers/users
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all
possible,
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='133'
global force_rport='No' peer/user force_rport='Yes')
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
'nat' for a peer/user that differs from the  global setting can make
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
peer/user discoverable by an attacker. Replies for non-existent
peers/users
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all
possible,
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='134'
global force_rport='No' peer/user force_rport='Yes')
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
'nat' for a peer/user that differs from the  global setting can make
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
peer/user discoverable by an attacker. Replies for non-existent
peers/users
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all
possible,
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='135'
global force_rport='No' peer/user force_rport='Yes')
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
'nat' for a peer/user that differs from the  global setting can make
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
peer/user discoverable by an attacker. Replies for non-existent
peers/users
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all
possible,
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! (config category='136'
global force_rport='No' peer/user force_rport='Yes')
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting
'nat' for a peer/user that differs from the  global setting can make
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that
peer/user discoverable by an attacker. Replies for non-existent
peers/users
[May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all
possible,
[May 27 15:52:33] WARNING[230

[asterisk-users] TDM800 just receive calls, but not make

[Vitor Mazuco]

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 14:21:31] WARNING[4332][C-000d]: chan_dahdi.c:13433
dahdi_request: Unknown option '-' in '6-1/29886874'
-- Called DAHDI/6-1/29886874
-- DAHDI/6-1 answered SIP/2000-000e
-- Channel DAHDI/6-1 joined 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
-- Channel SIP/2000-000e joined 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
> 0x7f2be401ce80 -- Probation passed - setting RTP source address to 
-- Channel SIP/2000-000e left 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
-- Channel DAHDI/6-1 left 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
== Spawn extension (ramais, 629886874, 1) exited non-zero on 'SIP/2000-000e'
-- Hanging up on 'DAHDI/6-1'
-- Hungup 'DAHDI/6-1'

and funny he gives answered right then or wait to ringing.

Out of nowhere he stopped, someone has been there?

Hugs

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Re: [asterisk-users] VoipRaider is true for FREE calls?

[Vitor Mazuco]

see the site here https://www.voipraider.com/calling_rates/

2016-05-09 19:43 GMT-03:00, Vitor Mazuco :
> 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 for a period of 90 days? Is that it? Has
> anyone tested it there? How many simultaneously calls can possible for
> Asterisk?
>

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[asterisk-users] VoipRaider is true for FREE calls?

[Vitor Mazuco]

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 for a period of 90 days? Is that it? Has
anyone tested it there? How many simultaneously calls can possible for
Asterisk?

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[asterisk-users] How to use Automatic Call Distribution for outbound calls with E1 links?

[Vitor Mazuco]

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

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Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

[Vitor Mazuco]

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 :
> On Wednesday 30 Mar 2016, Vitor Mazuco wrote:
>> Hi!
>>
>> Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or
>> any others digium card FXO for use Fax modem?
>
> Yes, in theory it is entirely possible to use an FXO card driven by software
>
> as a modem  (and indeed, this is exactly what Winmodems do);  although you
> will have to do all the hard work of generating the outgoing tones, and
> decoding the incoming tones, yourself.  This is a highly non-trivial task,
> and
> there is almost certain to be a better way than this of achieving whatever
> you
> want.
>
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
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[asterisk-users] Is possible to use FXO Digium card like a Fax modem?

[Vitor Mazuco]

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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Re: [asterisk-users] SIP trunk with whatsapp

[Vitor Mazuco]

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.
>
> https://github.com/tgalal/yowsup/issues/887
>
> Best regards.
>
> Emiliano.
>
>
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[asterisk-users] Can Chan Dongle using PPP for access to an IP network?

[Vitor Mazuco]

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 
https://uploaddeimagens.com.br/images/000/592/176/original/ras_vitor.png?1459178845

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Re: [asterisk-users] How to install Avaya 4610SW in Asterisk?

[Vitor Mazuco]

Humm ok

But is obrigated to install a FTP server in any host for this telephone
works?
Hi,

I've done a similar task (as a test) for a couple Avaya 1603 phones in my
office, and they also require a TFTP/HTTP server for the configuration
files.
Keep in mind, the server is only for those configuration files, and if you
are using a dhcp server for the phones, then you can define the server
address there, so in theory, the TFTP/HTTP Server can be installed on any
computer/server as long as the Avaya phones can access it.

In my case, i was using an AsteriskNow system, which comes bundled with
FreePBX and webserver, so my installation already had a HTTP server
running, and i just added the files there, and told my dhcp server the path
to it, and it worked fine.

Also, if i'm not mistaken, you have to convert the phones over to SIP
(Avaya usually defaults to H.323) unless you compiled H323 support into
your Asterisk installation.


TL;DR: No, it is not necessary to install the TFTP/HTTP server on your
Asterisk server, it can be installed anywhere, as long as your Avaya phones
can access it for their configuration files.

Kv. Birkir Freyr
Sími: 522-6069 / 896-6310


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] How to install Avaya 4610SW in Asterisk?

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.

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Sími +354 522 6000

www.vedur.is | en.vedur.is

E-mail Disclaimer<
http://www.vedur.is/um-vi/vefurinn/notkunarskilmalar/fyrirvari/>

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[asterisk-users] How to install Avaya 4610SW in Asterisk?

[Vitor Mazuco]

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.

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Re: [asterisk-users] How to recive Incoming calls in Chan Dongle ?

[Vitor Mazuco]

Humm yes, thanks very much !
Em 04/03/2016 18:00, "Ashish Gupta"  escreveu:

> Hi Vitor,
>
> The dongle.conf file contains your configuration setting related to your
> particular dongle. There, set the "context=dongle" (or anything you
> specified in extensions.conf), then provide the "exten=1234" (The
> extensions that will be called in the particular context). Also provide the
> imei number in the end of the file.
>
> Now in the extensions.conf file start with the context you provided in
> dongle.conf(eg. dongle). Now write the following command to answer the call
> coming through the dongle-
>
> [dongle]
> exten => 1234,1,Answer()
>
> HTH,
> Ashish
>
> *Ashish Gupta*
> *B.Tech (ECE) 3rd Year*
> *The LNM Institute of Information Technology*
> *Jaipur, Rajasthan - 302031 , India*
> *Mobile No: +917597056895 <%2B917597056895>*
>
> On Sat, Mar 5, 2016 at 2:15 AM, Vitor Mazuco 
> wrote:
>
>> 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
>>
>> pbx.c:6796 __ast_pbx_run: Channel 'Dongle/dongle1-01' sent to
>> invalid extension but no invalid handler:
>> context,exten,priority=URA,+5511965380290,1,Noop(),1
>>
>>
>> Thanks in advanced.
>>
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>
>
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[asterisk-users] How to recive Incoming calls in Chan Dongle ?

[Vitor Mazuco]

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

pbx.c:6796 __ast_pbx_run: Channel 'Dongle/dongle1-01' sent to
invalid extension but no invalid handler:
context,exten,priority=URA,+5511965380290,1,Noop(),1


Thanks in advanced.

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Re: [asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

[Vitor Mazuco]

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"  escreveu:

> On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco wrote:
>
> > I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but
> > my Huawei E153 is not working in my Asterisk.
> > But not successes.
>
>
> A little more information from you would be helpful to identify the
> problem.
>
> I have a Huawei USB 3G-stick and it works fine on Asterisk 11.
>
> Take a look here:
>
>
> http://www.raspberry-asterisk.org/documentation/gsm-voip-gateway-with-chan_dongle/comment-page-1/
>
> Not all Huawei USB modems work out of the box, on some of them voice
> calling capability has to be enabled first, some need to be upgraded
> with the latest firmware. Details on this can be found on the original
> chan_dongle wiki.
>
> https://github.com/bg111/asterisk-chan-dongle/wiki/Preparation
>
> Before inserting the SIM into your modem please deactivate the PIN on
> your card. This can be done with any phone. Insert the SIM into your
> phone, deactivate PIN and you’re done.
>
>
>
>
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[asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

[Vitor Mazuco]

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,

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[asterisk-users] Zoiper on Windows Phone

[Vitor Mazuco]

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
address to 179.XX.XXX.XX:57741
[Feb 29 12:32:28] NOTICE[4348][C-01ce]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '179.XX.XX.XX:57741'
   > 0x7efc90024190 -- Probation passed - setting RTP source
address to 179.XX.XX.XX:57741
-- Executing [1033@ramais:2] Set("SIP/1030-0201", "location=")
in new stack
-- Executing [1033@ramais:3] Verbose("SIP/1030-0201", "0,
getting push info  ") in new stack
 getting push info
-- Executing [1033@ramais:4] Set("SIP/1030-0201",
"regx="X-PUSH-URI=([0-9a-zA-Z\.\:\/\_]+)"") in new stack
-- Executing [1033@ramais:5] Set("SIP/1030-0201", "push=") in new stack
-- Executing [1033@ramais:6] System("SIP/1030-0201",
"/usr/bin/push.sh ") in new stack
-- Executing [1033@ramais:7] Wait("SIP/1030-0201", "1") in new stack
-- Executing [1033@ramais:8] Dial("SIP/1030-0201", "SIP/1033")
in new stack
[Feb 29 12:32:29] WARNING[4348][C-01ce]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/1030-0201' status is 'CHANUNAVAIL'
asterisk*CLI>

I've created the file more push.sh qualification in the dialplan. But the
Windows Phone can not run on Asterisk.

Does anyone know another method for this?

Thanks in advanced.

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Re: [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

[Vitor Mazuco]

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 :
> I tried this
>
> [dongle0]
> ;audio=/dev/ttyUSB1 ; tty port for audio connection;
>  no default value
> ;data=/dev/ttyUSB2  ; tty port for AT commands;
>  no default value
>
> ; or you can omit both audio and data together and use
> imei=123456789012345 and/or imsi=123456789012345
> ;  imei 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 :
>> On Fri, 2016-02-12 at 14:33 -0200, Vitor Mazuco wrote:
>>> Yes I used.
>>>
>>> The problem can be the version of Asterisk?
>>>
>>> I use Asterisk 13 instead of 11.
>>
>> Try
>>
>> [dongle0]
>> imei=347654458453667
>> imsi=976895757545778
>>
>>
>>
>> --
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Re: [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

[Vitor Mazuco]

I tried this

[dongle0]
;audio=/dev/ttyUSB1 ; tty port for audio connection;
 no default value
;data=/dev/ttyUSB2  ; tty port for AT commands;
 no default value

; or you can omit both audio and data together and use
imei=123456789012345 and/or imsi=123456789012345
;  imei 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 :
> On Fri, 2016-02-12 at 14:33 -0200, Vitor Mazuco wrote:
>> Yes I used.
>>
>> The problem can be the version of Asterisk?
>>
>> I use Asterisk 13 instead of 11.
>
> Try
>
> [dongle0]
> imei=347654458453667
> imsi=976895757545778
>
>
>
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Re: [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

[Vitor Mazuco]

Right,

I'll use in Asterisk 11 and I reply for you.

Thanks,

2016-02-12 14:35 GMT-02:00, Shabbir abbasi :
> 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, Vitor Mazuco 
> wrote:
>
>> Yes I used.
>>
>> The problem can be the version of Asterisk?
>>
>> I use Asterisk 13 instead of 11.
>>
>>
>>
>> 2016-02-12 14:31 GMT-02:00, Shabbir abbasi :
>> > have changed this
>> > [dongle0]
>> > audio=/dev/ttyUSB1
>> > data=/dev/ttyUSB2
>> >
>> > To
>> >
>> > [dongle0]
>> > imei=123456789012345
>> >
>> > and imei exact same as on your device ?
>> >
>> > On Fri, Feb 12, 2016 at 9:29 PM, Vitor Mazuco 
>> > wrote:
>> >
>> >> Yes, I used IMEI.
>> >>
>> >> But in CLI appearing nothing and it not register.
>> >>
>> >>
>> >>
>> >> 2016-02-12 14:27 GMT-02:00, Shabbir abbasi
>> >> :
>> >> > have you tried   imei discovery
>> >> > imei=123456789012345
>> >> >
>> >> >
>> >> > write imei number instaed of 12345...
>> >> >
>> >> > On Fri, Feb 12, 2016 at 8:51 PM, Vitor Mazuco
>> >> > > >
>> >> > wrote:
>> >> >
>> >> >> 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
>> >> >> -- [dongle0] Trying to connect on /dev/ttyUSB1...
>> >> >> -- [dongle0] Dongle has connected, initializing...
>> >> >> [Feb 12 13:49:25] ERROR[13348]: chan_dongle.c:442 do_monitor_phone:
>> >> >> [dongle0] timedout while waiting 'OK' in response to 'AT'
>> >> >> -- [dongle0] Error initializing Dongle
>> >> >> -- [dongle0] Dongle has disconnected
>> >> >>
>> >> >>
>> >> >> In dongle.conf I use
>> >> >>
>> >> >> [dongle0]
>> >> >> audio=/dev/ttyUSB1
>> >> >> data=/dev/ttyUSB2
>> >> >>
>> >> >> Somebody already uses this software?
>> >> >>
>> >> >> Thanks in advanced.
>> >> >>
>> >> >> --
>> >> >> _
>> >> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com
>> --
>> >> >> New to Asterisk? Join us for a live introductory webinar every
>> >> >> Thurs:
>> >> >>http://www.asterisk.org/hello
>> >> >>
>> >> >> asterisk-users mailing list
>> >> >> To UNSUBSCRIBE or update options visit:
>> >> >>http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >>
>> >> >
>> >>
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>> >>http://www.asterisk.org/hello
>> >>
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>> >> To UNSUBSCRIBE or update options visit:
>> >>http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >
>>
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Re: [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

[Vitor Mazuco]

Yes I used.

The problem can be the version of Asterisk?

I use Asterisk 13 instead of 11.



2016-02-12 14:31 GMT-02:00, Shabbir abbasi :
> have changed this
> [dongle0]
> audio=/dev/ttyUSB1
> data=/dev/ttyUSB2
>
> To
>
> [dongle0]
> imei=123456789012345
>
> and imei exact same as on your device ?
>
> On Fri, Feb 12, 2016 at 9:29 PM, Vitor Mazuco 
> wrote:
>
>> Yes, I used IMEI.
>>
>> But in CLI appearing nothing and it not register.
>>
>>
>>
>> 2016-02-12 14:27 GMT-02:00, Shabbir abbasi :
>> > have you tried   imei discovery
>> > imei=123456789012345
>> >
>> >
>> > write imei number instaed of 12345...
>> >
>> > On Fri, Feb 12, 2016 at 8:51 PM, Vitor Mazuco 
>> > wrote:
>> >
>> >> 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
>> >> -- [dongle0] Trying to connect on /dev/ttyUSB1...
>> >> -- [dongle0] Dongle has connected, initializing...
>> >> [Feb 12 13:49:25] ERROR[13348]: chan_dongle.c:442 do_monitor_phone:
>> >> [dongle0] timedout while waiting 'OK' in response to 'AT'
>> >> -- [dongle0] Error initializing Dongle
>> >> -- [dongle0] Dongle has disconnected
>> >>
>> >>
>> >> In dongle.conf I use
>> >>
>> >> [dongle0]
>> >> audio=/dev/ttyUSB1
>> >> data=/dev/ttyUSB2
>> >>
>> >> Somebody already uses this software?
>> >>
>> >> Thanks in advanced.
>> >>
>> >> --
>> >> _
>> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> >>http://www.asterisk.org/hello
>> >>
>> >> asterisk-users mailing list
>> >> To UNSUBSCRIBE or update options visit:
>> >>http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >
>>
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>> _
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>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>

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Re: [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

[Vitor Mazuco]

Yes, I used IMEI.

But in CLI appearing nothing and it not register.



2016-02-12 14:27 GMT-02:00, Shabbir abbasi :
> have you tried   imei discovery
> imei=123456789012345
>
>
> write imei number instaed of 12345...
>
> On Fri, Feb 12, 2016 at 8:51 PM, Vitor Mazuco 
> wrote:
>
>> 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
>> -- [dongle0] Trying to connect on /dev/ttyUSB1...
>> -- [dongle0] Dongle has connected, initializing...
>> [Feb 12 13:49:25] ERROR[13348]: chan_dongle.c:442 do_monitor_phone:
>> [dongle0] timedout while waiting 'OK' in response to 'AT'
>> -- [dongle0] Error initializing Dongle
>> -- [dongle0] Dongle has disconnected
>>
>>
>> In dongle.conf I use
>>
>> [dongle0]
>> audio=/dev/ttyUSB1
>> data=/dev/ttyUSB2
>>
>> Somebody already uses this software?
>>
>> Thanks in advanced.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>

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[asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

[Vitor Mazuco]

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
-- [dongle0] Trying to connect on /dev/ttyUSB1...
-- [dongle0] Dongle has connected, initializing...
[Feb 12 13:49:25] ERROR[13348]: chan_dongle.c:442 do_monitor_phone:
[dongle0] timedout while waiting 'OK' in response to 'AT'
-- [dongle0] Error initializing Dongle
-- [dongle0] Dongle has disconnected


In dongle.conf I use

[dongle0]
audio=/dev/ttyUSB1
data=/dev/ttyUSB2

Somebody already uses this software?

Thanks in advanced.

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[asterisk-users] WhatsApp VoIP in Asterisk integration?

[Vitor Mazuco]

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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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

Humm thanks very much :)
Em 04/02/2016 19:58, "Doug Lytle"  escreveu:

>
>
> >>> On Feb 4, 2016, at 12:55 PM, Vitor Mazuco vitor.maz...@gmail.com
> wrote:
>
> >>> so this context parkedcalls is inside on features.conf?
>
> Correct.
>
> Doug
>
>
>
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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

Humm,
so this context parkedcalls is inside on features.conf?

2016-02-03 17:42 GMT-02:00, Doug Lytle :
>>>> 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 parkedcalls?
>
> Nothing,
>
> The context parkedcalls is generated by features.conf, you just need to
> include it in your dialplan
>
> CLI> dialplan show parkedcalls
>
> [ Context 'parkedcalls' created by 'features' ]
>   '700' =>  1. Park()
> [features]
>
>
> Doug
>
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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

Ah no, I'm asking what code I put inside of parkedcalls?

This example works?

[ramais]

include => parkedcalls

[parkedcalls]
exten => 700,1,ParkedCall(701)
exten => 702,1,ParkedCall(702)
exten => 703,1,ParkedCall(703)
exten => 704,1,ParkedCall(704)

This exten works?


2016-02-03 17:27 GMT-02:00, Doug Lytle :
>>>> 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 you give an example?
>
>
> [ramais]
>
> include => parkedcalls
>
>
> Doug
>
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Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

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 :
> On Wed, Feb 3, 2016 at 1:05 PM, Vitor Mazuco 
> wrote:
>
>> 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'
>>
>
> Are you loading res_parking.so?
>
> Does your res_parking.conf define a parkext and specify the context?
> Documented in configs/samples/res_parking.conf.sample:
> parkext => 700  ; What extension to dial to park.
> (optional; if
> ; specified, extensions will be created for
> parkext and
> ; the whole range of parkpos)
> context => parkedcalls  ; Which context parked calls and the
> default park
>
> Once that is configured you can include the parkedcalls context into
> your ramais context.
>
> Richard
>

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[asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

[Vitor Mazuco]

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?

Can be this example below?

[parkedcalls]
exten => 700,1,ParkedCall(701)
exten => 702,1,ParkedCall(702)
exten => 703,1,ParkedCall(703)
exten => 704,1,ParkedCall(704)

If not, somebody knows that?

Thanks

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[asterisk-users] cdr_odbc: Error in ExecDirect: -1

[Vitor Mazuco]

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] Dial("SIP/2020-",
"SIP/2021,60,tT") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/2021
-- SIP/2021-0001 is ringing
   > 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
[Jan 13 11:31:07] NOTICE[2279][C-]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '(null)'
-- SIP/2021-0001 answered SIP/2020-
   > 0x7fd3b4004eb0 -- Probation passed - setting RTP source
address to 192.168.25.100:8000
   > 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
   > cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] WARNING[2279][C-]: res_odbc.c:604
ast_odbc_direct_execute: SQL Execute error! Verifying connection to
asterisk [asterisk-connector]...
   > cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] ERROR[2279][C-]: cdr_odbc.c:177 odbc_log:
CDR direct execute failed


See my res_odbc.conf

[asterisk]
enabled = yes
dsn = asterisk-connector
username = root
password = 100567
pooling = no
limit = 1
pre-connect = yes

What can be happened?

Thank in advanced.

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[asterisk-users] The To header was truncated in call... Whats this means?

[Vitor Mazuco]

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 '18e0a12e434364254b0cc2e52d20755b@191.x. This call setup
will fail.
...

Whats this massege means?

Thanks.

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Re: [asterisk-users] Detected alarm on channel 3: Red Alarm

[Vitor Mazuco]

Oh, thanks very much Shaun,

I'm from Brazil

And about the voltage, how can I fix this red alarm?
Em 05/01/2016 13:23, "Shaun Ruffell"  escreveu:

> On Tue, Jan 05, 2016 at 02:36:42PM +, Ryan, Travis wrote:
> > > -Original Message-
> > > 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
> > > Subject: [asterisk-users] Detected alarm on channel 3: Red Alarm
> > >
> > > 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.
> > >
> > > What can be happen?
> > >
> > > My lines is all ok! But when I put on Digium Card TDM410 is very
> > > inconsistent
> > >
> > > Thanks
> > >
> >
> > Honestly, I've only had red alarms on any of my cards if there was
> > a problem with the lines or service over those lines. Maybe
> > someone else could speak to other reasons the red light might
> > appear.
>
> The red alarm on the analog cards, like the TDM410, appear when the
> card does not detect a sufficient battery voltage from the central
> office.
>
> Typically this will happen when the cable is disconnected from the
> card, there is something flaky in the connection with the central
> office, a filter is causing a voltage drop, etc..
>
> One thing to check is that the 'opermode' module parameter is set
> for the country that you're in, if you're not in the United States
> or Canada.
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
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Re: [asterisk-users] Detected alarm on channel 3: Red Alarm

[Vitor Mazuco]

My line is comming from a monden ADSL that it provide internet too.



2016-01-05 12:46 GMT-02:00, Ryan, Travis :
>
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Vitor Mazuco
>> Sent: Tuesday, January 05, 2016 9:42 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Detected alarm on channel 3: Red Alarm
>>
>> Humm, if I put a filter in this lines, maybe back?
>>
>>
>>
>> 2016-01-05 12:36 GMT-02:00, Ryan, Travis :
>> >
>> >> -Original Message-
>> >> 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
>> >> Subject: [asterisk-users] Detected alarm on channel 3: Red Alarm
>> >>
>> >> 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.
>> >>
>> >> What can be happen?
>> >>
>> >> My lines is all ok! But when I put on Digium Card TDM410 is very
>> >> inconsistent
>> >>
>> >> Thanks
>> >>
>> >> --
>> > Honestly, I've only had red alarms on any of my cards if there was a
>> > problem with the lines or service over those lines. Maybe someone
>> else
>> > could speak to other reasons the red light might appear.
>> >
>> >
>> > --
>
> [Ryan, Travis] Is your line pretty long? Maybe the distance is far so the
> signal is weak?
>
>
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Re: [asterisk-users] Detected alarm on channel 3: Red Alarm

[Vitor Mazuco]

Humm, if I put a filter in this lines, maybe back?



2016-01-05 12:36 GMT-02:00, Ryan, Travis :
>
>> -Original Message-
>> 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
>> Subject: [asterisk-users] Detected alarm on channel 3: Red Alarm
>>
>> 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.
>>
>> What can be happen?
>>
>> My lines is all ok! But when I put on Digium Card TDM410 is very
>> inconsistent
>>
>> Thanks
>>
>> --
> Honestly, I've only had red alarms on any of my cards if there was a problem
> with the lines or service over those lines. Maybe someone else could speak
> to other reasons the red light might appear.
>
>
> --
> _
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

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[asterisk-users] Detected alarm on channel 3: Red Alarm

[Vitor Mazuco]

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.

What can be happen?

My lines is all ok! But when I put on Digium Card TDM410 is very inconsistent

Thanks

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Re: [asterisk-users] Best Asterisk Platform

[Vitor Mazuco]

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 through a
> contractor.
>
> I dont mind paying a license fee for a front end interface but still would
> rather not have to pay.
>
> Thanks,
> --Eric
>

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[asterisk-users] Is possible to install CDR-Viewer in Asterisk 11?

[Vitor Mazuco]

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 retrieve database
handle CDR failed."

See the full log

[/code]
cdr-teste*CLI> module reload res_odbc.so
-- Reloading module 'res_odbc.so' (ODBC resource)
  == Parsing '/etc/asterisk/res_odbc.conf': Found
[Dec 14 10:44:03] NOTICE[1591]: res_odbc.c:1529 odbc_obj_connect:
Connecting asterisk
[Dec 14 10:44:03] NOTICE[1591]: res_odbc.c:1568 odbc_obj_connect:
res_odbc: Connected to asterisk [asterisk-connector]
[Dec 14 10:44:03] NOTICE[1591]: res_odbc.c:919 load_odbc_config:
Registered ODBC class 'asterisk' dsn->[asterisk-connector]
cdr-teste*CLI>
  == Using SIP RTP CoS mark 5
-- Executing [2020@ramais:1] Dial("SIP/2021-0002",
"SIP/2020,60,tT") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/2020
-- SIP/2020-0003 is ringing
   > 0xb6e2f110 -- Probation passed - setting RTP source address
to 192.2.1.165:57040
[Dec 14 10:44:16] NOTICE[1593][C-0001]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '(null)'
-- SIP/2020-0003 answered SIP/2021-0002
   > 0xb6e10210 -- Probation passed - setting RTP source address
to 192.2.1.60:8000
   > 0xb6e2f110 -- Probation passed - setting RTP source address
to 192.2.1.165:57040
[Dec 14 10:44:17] ERROR[1593][C-0001]: cdr_odbc.c:163 odbc_log:
Unable to retrieve database handle.  CDR failed.
  == Spawn extension (ramais, 2020, 1) exited non-zero on 'SIP/2021-0002'
[/code]

I don't know if can be the problem in my files of configuration

[code]/etc/asterisk/res_odbc.conf[/code]

[code]
[asterisk]
enabled => yes
dsn => asterisk-connector
username => root
password => 100567
pooling => no
limit => 1
pre-connect => yes
[/code]

And my [code]/etc/asterisk/cdr_odbc.conf[/code]

[code]
;
; cdr_odbc.conf
;
[global]
dsn=asterisk-connector
loguniqueid=yes
username=asterisk
password=100567
dispositionstring=yes
table=cdr   ;"cdr" is default table name
usegmtime=no ; set to "yes" to log in GMT
hrtime=yes  ;Enables microsecond accuracy with the billsec
and duration fields
;newcdrcolumns=yes ; Enable logging of post-1.8 CDR columns
(peeraccount, linkedid, sequence)
~
[/code]

Can anyone tell me what is the problem?

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Re: [asterisk-users] Help with CDR-Stats

[Vitor Mazuco]

Right, thanks for your reply!



2015-12-16 14:45 GMT-02:00, Bruce Ferrell :
> billing is sending invoices for calls to customers.
>
> reporting is overall statistics on the aggregate of your calls...
> Average call hold time, common (or uncommon destinations) etc.  If you
> see a destination that suddenly has a lot of calls with hold time below
> normal, there may be a call quality problem.
>
> TPC (the phone company) has used statistical troubleshooting techniques
> for decades to keep quality up so customers don't have to complain, 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" > <mailto: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!
>>
>> I'm trying to install CDR-Stats (cdr-stats.org
>> <http://cdr-stats.org>), but it very difficult.
>>
>> Is there others optins for billing?
>>
>> Thanks
>>
>>
>>
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>>
>>
>>
>
>

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Re: [asterisk-users] Help with CDR-Stats

[Vitor Mazuco]

Humm whats is the diferent?
Em 16/12/2015 14:19, "Annus Fictus"  escreveu:

> CDR-STATS is for reporting.
>
> A2Billing is for billing...
>
> Regards
>
> El 16/12/2015 a las 11:15, Vitor Mazuco escribió:
>
>> 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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[asterisk-users] Help with CDR-Stats

[Vitor Mazuco]

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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