[asterisk-users] choppy calls version Asterisk 13.17 on CentOS 7

[Motty Cruz]

Hello, 

A few months ago we upgraded our server from Asterisk 1.8.22.0 on CentOS 5.9
to Asterisk 13.13.1 on CentOS 7. We are still using SIP not PJSIP.  Since
the upgrade our remote users' conversions are choppy.  Here is what my
sip.conf looks like for the users with most problems: 

 

[user-name]

type=friend

context=sip-phone

trustrpid=no

call-limit=2

callerid="Mike" <1008>

disallow=all

allow=g726

allow=g722

allow=g723

allow=ulaw

allow=alaw

username=user-name

secret=

dtmfmode=rfc2833

host=dynamic

mailbox=1008

nat=force_rport,comedia

canreinvite=no

 

any ideas? Monitoring using CLI, I noticed the device always select ulaw for
codec.  

 

Thanks, 
Motty

 

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[asterisk-users] upgrading asterisk 13.13.1 to latest version best practices

[Motty Cruz]

Hello, 

Best practices examples to upgrade Asterisk 13.13.1 to latest version?  

 

Any suggestions? 

 

Thanks,
Motty

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Re: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

[Motty Cruz]

Hello, I am trying to match SRC in CSV file to clid - Caller ID to user
extension number for stats purposes, however, in CSV file the SRC is the
company number as set in Extensions.conf 

 

exten => _7XXX,1,Set(CALLERID(number)="Company Inc" <3788001800>)

exten => _7XXX,2,Dial(SIP/voip1/13781${EXTEN:1},80)

exten => _7XXX,n,Congestion()

exten => _7XXX,n,Hangup()

 

how would I change it? I have look in cdr.conf and logger.conf

 

Thanks, 

 

From: Motty Cruz [mailto:motty.c...@gmail.com] 
Sent: Monday, April 03, 2017 3:52 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: motty.c...@gmail.com
Subject: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID
as CallerID

 

Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to  log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number is 2318001800 which is the main company's number set in
Extensions.conf. 

 



2318001800

78079745

phones

"ITadmin" <2318001800>

SIP/4007-00015c0a

SIP/voip1-00015c0b

Dial

SIP/voip1/78079745,80

4/3/2017 15:30

4/3/2017 15:31

2

0

NO ANSWER

DOCUMENTATION

1.49E+09




 

 

Logger.conf

[general]

dateformat=%F %T

;

; Customize the display of debug message time stamps

; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)

;

; see strftime(3) Linux manual for format specifiers.  Note that there is
also

; a fractional second parameter which may be used in this field.  Use %1q

; for tenths, %2q for hundredths, etc.

;

;dateformat=%F %T   ; ISO 8601 date format

;dateformat=%F %T.%3q   ; with milliseconds

dateformat = %F %T.%3q   ; ISO 8601 date format with milliseconds

;

;

; This makes Asterisk write callids to log messages

; (defaults to yes)

use_callids = yes

 

 

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[asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

[Motty Cruz]

Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to  log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number is 2318001800 which is the main company's number set in
Extensions.conf. 

 



2318001800

78079745

phones

"ITadmin" <2318001800>

SIP/4007-00015c0a

SIP/voip1-00015c0b

Dial

SIP/voip1/78079745,80

4/3/2017 15:30

4/3/2017 15:31

2

0

NO ANSWER

DOCUMENTATION

1.49E+09




 

 

Logger.conf

[general]

dateformat=%F %T

;

; Customize the display of debug message time stamps

; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)

;

; see strftime(3) Linux manual for format specifiers.  Note that there is
also

; a fractional second parameter which may be used in this field.  Use %1q

; for tenths, %2q for hundredths, etc.

;

;dateformat=%F %T   ; ISO 8601 date format

;dateformat=%F %T.%3q   ; with milliseconds

dateformat = %F %T.%3q   ; ISO 8601 date format with milliseconds

;

;

; This makes Asterisk write callids to log messages

; (defaults to yes)

use_callids = yes

 

 

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[asterisk-users] how to hangup this channel "Message/ast_msg_queu

[Motty Cruz]

omega*CLI> core show channels
Channel  Location State Application(Data)
Message/ast_msg_queu 4002@sipphones:2 Up VoiceMail(4002@default,u)

"Message/ast_msg_queu" it's been up for the last day, how to hangup this 
channel?



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Re: [asterisk-users] CDR reporting solution

[motty cruz]

I installed CDR-Stats on Debian 8.7
http://cdr-stats.readthedocs.io/en/latest/installation/install-cdr-stats.html

I am trying to figure out how to import flat CSV file to CDR-Stats



On Wed, Mar 22, 2017 at 6:28 PM, Bruce Ferrell <bferr...@baywinds.org>
wrote:

> How about CDR to either MySQL or cdrlite and a quickie sql query? I could
> have added postgres, but I'm a DB bigot.  That would work too.
>
>
> On 03/22/2017 01:46 PM, Motty Cruz wrote:
>
>>
>> Hello, I am looking for CDR reporting solution? Any suggestions? I am
>> using Asterisk 13.13.1
>>
>> I would like a report on number of calls per extension.
>>
>> Thanks,
>> Motty
>>
>>
>>
>>
>
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Motty
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[asterisk-users] CDR reporting solution

[Motty Cruz]

Hello, I am looking for CDR reporting solution? Any suggestions? I am using
Asterisk 13.13.1

I would like a report on number of calls per extension. 

 

Thanks, 
Motty

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[asterisk-users] fail2ban Asterisk 13.13.1

[Motty Cruz]

Hello, fail2ban does not ban offending IP. 

 

NOTICE[29784] chan_sip.c: Registration from
'"user3"' failed for 'offending-IP:53417' - Wrong
password

NOTICE[29784] chan_sip.c: Registration from
'"user3"' failed for ‘offending-IP:53911' -
Wrong password

 

systemctl status fail2ban

● fail2ban.service - Fail2Ban Service

   Loaded: loaded (/usr/lib/systemd/system/fail2ban.service; enabled; vendor
preset: disabled)

   Active: active (running) since Wed 2017-03-01 00:40:43 PST; 470min ago

 Docs: man:fail2ban(1)

 

jail.local

[DEFAULT]

# "bantime" is the number of seconds that a host is banned.

bantime  = -1

 

# A host is banned if it has generated "maxretry" during the last "findtime"

# seconds.

findtime  = 300

 

# "maxretry" is the number of failures before a host get banned.

maxretry = 3

 

[asterisk-iptables]

enable = true

port = 5060,5061

filter   = asterisk

action   = iptables-allports[name=ASTERISK, protocol=all]

  sendmail[name=ASTERISK, dest=mo...@email.com,
sender=fail2...@asterisk-ip.com]

#action   = %(banaction)s[name=%(__name__)s-tcp, port="%(port)s",
protocol="tcp", chain="%(chain)s", actname=%(banaction)s-tcp]

   %(banaction)s[name=%(__name__)s-udp, port="%(port)s",
protocol="udp", chain="%(chain)s", actname=%(banaction)s-udp]

   %(mta)s-whois[name=%(__name__)s, dest="%(destemail)s"]

logpath  = /var/log/asterisk/messages

maxretry = 3

findtime  = 300

bantime  = -1

 

 

in filter.d

asterisk.conf

failregex = ^%(__prefix_line)s%(log_prefix)s Registration from '[^']*'
failed for '(:\d+)?' - (Wrong password|Username/auth name mismatch|No
matching peer found|Not a local domain|Device does not match ACL|Peer is not
supposed to register|ACL error \(permit/deny\)|Not a local domain)$

^%(__prefix_line)s%(log_prefix)s Call from '[^']*'
\(:\d+\) to extension '[^']*' rejected because extension not found in
context

^%(__prefix_line)s%(log_prefix)s Host  failed to
authenticate as '[^']*'$

^%(__prefix_line)s%(log_prefix)s No registration for peer
'[^']*' \(from \)$

^%(__prefix_line)s%(log_prefix)s Host  failed MD5
authentication for '[^']*' \([^)]+\)$

^%(__prefix_line)s%(log_prefix)s Failed to authenticate
(user|device) [^@]+@\S*$

^%(__prefix_line)s%(log_prefix)s hacking attempt detected
''$

^%(__prefix_line)s%(log_prefix)s
SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPa
ssword)",EventTV="([\d-]+|%(iso8601)s)",Severity="[\w]+",Service="[\w]+",Eve
ntVersion="\d+",AccountID="(\d*|)",SessionID=".+",LocalAddress="IPV
[46]/(UDP|TCP|WS)/[\da-fA-F:.]+/\d+",RemoteAddress="IPV[46]/(UDP|TCP|WS)//\d+"(,Challenge="[\w/]+")?(,ReceivedChallenge="\w+")?(,Response="\w+",Ex
pectedResponse="\w*")?(,ReceivedHash="[\da-f]+")?(,ACLName="\w+")?$

^%(__prefix_line)s%(log_prefix)s "Rejecting unknown SIP
connection from "$

^%(__prefix_line)s%(log_prefix)s Request (?:'[^']*' )?from '[^']
*' failed for '(?::\d+)?'\s\(callid: [^\)]*\) - (?:No matching
endpoint found|Not match Endpoint(?: Contact)? ACL|(?:Failed|Error) to
authenticate)\s*$

 

failregex = NOTICE.* .*: Registration from '.*' failed for '' - Wrong
password

NOTICE.* .*: Registration from '.*' failed for ':.*' - No
matching peer found

NOTICE.* .*: Registration from '.*' failed for '' - No
matching peer found

NOTICE.* .*: Registration from '.*' failed for '' -
Username/auth name mismatch

NOTICE.* .*: Registration from '.*' failed for '' - Device
does not match ACL

NOTICE.* .*: Registration from '.*' failed for '' - Peer
is not supposed to register

NOTICE.* .*: Registration from '.*' failed for '' - ACL
error (permit/deny)

NOTICE.* .*: Registration from '.*' failed for '' - Device
does not match ACL

NOTICE.*  failed to authenticate as '.*'$

NOTICE.* .*: No registration for peer '.*' \(from \)

NOTICE.* .*: Host  failed MD5 authentication for '.*' (.*)

NOTICE.* .*: Failed to authenticate user .*@.*

NOTICE.* .*: Sending fake auth rejection for device
.*\;tag=.*

NOTICE.* .*: Registration from '\".*\".*' failed for '' -
No matching peer found

NOTICE.* .*: Registration from '\".*\".*' failed for '' -
Wrong password

 

ignoreregex =

 

Thanks

Motty

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[asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Motty Cruz]

Hello, I have a user that prefers Soft SIP phone install on his laptop, for
security reasons I have enable TLS on our Asterisk server to support TLS
authentication, It works well with hard phones. Has anybody in this forum
use SIP Soft phones with TLS authentication enabled? Any suggestions? 

 

Thanks, 
Motty

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[asterisk-users] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)

[Motty Cruz]

Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error: 

 

-- SIP/voipeer-084b redirecting info has changed, passing it to
SIP/1007-084a

-- SIP/voipeer-084b is busy

  == Everyone is busy/congested at this time (1:1/0/0)

-- Timeout on SIP/1007-084a

-- Executing [t@phones:1] Playback("SIP/1007-084a", "goodbye") in
new stack

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

--  Playing 'goodbye.slin' (language 'en')

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

-- Executing [t@phones:2] Hangup("SIP/1007-084a", "") in new stack

 

Sip.conf 

[1007]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1007>
disallow=all
allow=ulaw
allow=alaw
username=1007
secret=X
dtmfmode=rfc2833
host=dynamic
mailbox=1007@default
nat=force_rport,comedia

 

Is it a codec issue? Or missed configuration? Asterisk does not know how to
translate busy signal. 

Your help is appreciated!

Thanks,

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Re: [asterisk-users] Asterisk 13.13.1

[Motty Cruz]

I thought it was a firewall issues. I disabled IP Tables & Selinux, but the
problem persist! I have not made changes on our firewall since the upgrade!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
>>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from
here:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar
.gz 

>>> I continue to see errors like this: 
>>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
Retransmission timeout reached on transmission
56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request)
-- See >>> >>> 

Firewall?

Doug

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Re: [asterisk-users] Asterisk 13.13.1

[Motty Cruz]

Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: 
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz 
  

 

I continue to see errors like this: 

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
6e3dd238-911e2ac3-f1260152@192.168.125.152 for seqno 103 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
ef497d11-a81b1c00-8bfbd3bf@192.168.1.244 for seqno 103 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

 

Before upgrading to this new server, Asterisk version 1.8 on  CentOS 5.9 
hardware on both servers were similar in CPU, Memory 

 

Any support on this matter is appreciated!

 

Thanks, 
Motty   

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kambiz sharifi
Sent: Saturday, January 28, 2017 5:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1

 

 

On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote:

What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
How did you installed Asterisk 1.8 and 13 ? From source or from package ?

I would be curious to see what would happen after downgrading back to 1.8.

 

2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>:

Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are 
starting to complaint about packets loss, conversations are choppy!  

 

PkEI don’t even know where to start looking! Choppy conversations happened 
within users. I am using sip.conf

 

[1091]

type=friend

context=sip-phone

call-limit=2

trustrpid=no

callerid="dev1" <1091>

disallow=all

allow=ulaw

allow=alaw

username=1091

secret=X

dtmfmode=rfc2833

host=dynamic

mailbox=10091@default

nat=force_rport,comedia

canreinvite=no

 

extensions.conf

exten => 1091,hint,SIP/${EXTEN}

exten => 1091,1,Dial(SIP/${EXTEN},15,t)

exten => 1091,2,Voicemail(${EXTEN}@default,u)

exten => 1091,102,Voicemail(${EXTEN}@default,b)

exten => 1091,103,Hangup

 

[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: 

Retransmission timeout reached on transmission 
7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

 

any ideas? 

 

Thanks!

Motty


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[asterisk-users] Asterisk 13.13.1

[Motty Cruz]

Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!


 

I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf

 

[1091]

type=friend

context=sip-phone

call-limit=2

trustrpid=no

callerid="dev1" <1091>

disallow=all

allow=ulaw

allow=alaw

username=1091

secret=X

dtmfmode=rfc2833

host=dynamic

mailbox=10091@default

nat=force_rport,comedia

canreinvite=no

 

extensions.conf

exten => 1091,hint,SIP/${EXTEN}

exten => 1091,1,Dial(SIP/${EXTEN},15,t)

exten => 1091,2,Voicemail(${EXTEN}@default,u)

exten => 1091,102,Voicemail(${EXTEN}@default,b)

exten => 1091,103,Hangup

 

[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: 

Retransmission timeout reached on transmission
7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

 

any ideas? 

 

Thanks!

Motty

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Re: [asterisk-users] T1 -Asterisk server - Analog lines

[Motty Cruz]

Thanks for your suggestion! After some research I purchased the AdTran Total
Access 904 2nd. We have a T1 connection to our Asterisk server and from my
Asterisk server to AdTran. I am in dead in the water, I can’t figure out how
to configure AdTran to talk to the Asterisk server.  

 

PRI à Asterisk àAdTran (total access 904 2nd gen)

 

Calls are being forward to AdTran from Asterisk!

 

Thanks, 
Motty

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, December 06, 2016 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 -Asterisk server - Analog lines

 

On 12/06/2016 06:45 PM, Motty Cruz wrote:

Any suggestions on how to convert digital signal to analog? 


I do this with an ADIT 600 channel bank that I got off of ebaY many years
ago.  A quick search shows one for around $98

http://www.ebay.com/itm/like/191565965269?lpid=82
<http://www.ebay.com/itm/like/191565965269?lpid=82=ps_noapp=true>
=ps_noapp=true

Doug

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Re: [asterisk-users] how to add area code to outgoing number in Asterisk 13.13

[Motty Cruz]

Thank you Carlos, you’re right I am using PJSIP. Should I not use it? 

 

Thanks, 
Motty

 

From: Carlos Chavez [mailto:cur...@telecomab.mx] 
Sent: Saturday, December 31, 2016 5:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Motty Cruz
Subject: Re: [asterisk-users] how to add area code to outgoing number in 
Asterisk 13.13

 

On 2016-12-29 12:44, Motty Cruz wrote:

Hi, my SIP provider requires 10 digits for all outgoing calls; Users dial 7 
digits for outgoing.  Here is how I added the area code to all outgoing calls 
in Asterisk 1.8

 

Extensions.conf

; Adding Area code and striping 7 for local numbers

exten => _7XXX,n,Set(CALLERID(all)="My ID" )

exten => _7XXX,n,Dial(SIP/mySIPprovider/1731${EXTEN:1},80)

 

This syntax does not work in Asterisk 13.13. has anybody dealt with this issue? 

 

Thanks
Motty

 

 

The syntax has not changed so it should work.  You are still using chan_sip 
right?  Maybe you are using PJSIP which does have a different syntax?  It is 
probably better if you send the output from the cli so we can see what error 
the dialplan is throwing at you.

-- 

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Carlos Chávez
dCAP #1349
+52 (55)8116-9161

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[asterisk-users] how to add area code to outgoing number in Asterisk 13.13

[Motty Cruz]

Hi, my SIP provider requires 10 digits for all outgoing calls; Users dial 7
digits for outgoing.  Here is how I added the area code to all outgoing
calls in Asterisk 1.8

 

Extensions.conf

; Adding Area code and striping 7 for local numbers

exten => _7XXX,n,Set(CALLERID(all)="My ID" )

exten => _7XXX,n,Dial(SIP/mySIPprovider/1731${EXTEN:1},80)

 

This syntax does not work in Asterisk 13.13. has anybody dealt with this
issue? 

 

Thanks
Motty

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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Motty Cruz]

Yves, 

I have a SoundStation IP 6000:

My sip.conf

[1006]

type=friend

context=sip-phone

call-limit=1

trustrpid=no

callerid="Conference Room3" <1006>

disallow=all

allow=ulaw

allow=alaw

username=1006

secret=secret1

dtmfmode=rfc2833

host=dynamic

mailbox=1000

nat=yes

canreinvite=no

 

Asterisk server IP XX.XX.42.16 (Public)

Client IP: 192.168.1.56

 

I would look for typos in your configuration!

 

-Motty

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, December 22, 2016 6:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

 

Do you have any LLDP or CDP enabled anywhere ?

 

2016-12-21 19:50 GMT+01:00 Victor Villarreal :

Hi Yves,

Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of the 
phone. Maybe with the snom this not happen because your switch don't see the 
MAC of the Snom as a "supperted IP Phone".

 

2016-12-21 13:59 GMT-03:00 Yves :

sorry... typo
the problematic phone has the 192.168.0.13
the asterisk has 192.168.1.211

when i connect a snom phone on the cable that was in the soundstation 6000 
before and configure the
phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...

it would be helpful if someone, that has a running soundstation ip 6000 could 
send the configuration... :-/

regards,
yves




Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:

On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:

Hi Mark,

yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config... remember...
when I use tcp the phone tries to register, but does not even try with
udp...

thank you,
yves

   I am a bit confused: is your problematic phone's IP 192.168.0.13
(what the error log is reporting below) or 192.168.1.13?

Am 21.12.2016 um 13:34 schrieb Mark Wiater:

Yves,

Didn't you say that

AsteriskServer: 192.168.1.211
SIP-user: 165

?

On 12/21/2016 4:24 AM, Yves wrote:

. It is sure for 100% that there is no firewall or something else mangeling
in between... another Hardphone works as expected using the same
Netzworkcable on the same Networkplug with UDP on Port 5060...


This other hardphone, what IP does it have?


50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
255.255.255.0

The line above suggests to me that your phone and your asterisk server are
on a different network, there has to be something that routes between those
two networks. Often what routes, can firewall.

000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
Temporarily not available



Mark




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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Motty Cruz]

can you provide the configuration on sip.conf file? Do you have the following 
settings under the account number or ext number? 

 

host=dynamic

nat=yes

 

for instance my configuration sip.conf file is as follow: 

[1005]

type=friend

context=sip-phone

call-limit=1

trustrpid=no

callerid="iuser 1005"

disallow=all

allow=ulaw

allow=alaw

username=1005

auth=md5

secret=819c8ebd2d1525235235325235

dtmfmode=rfc2833

host=dynamic

nat=yes

canreinvite=no

 

Thanks, 
Motty

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, December 19, 2016 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

 

 

 

2016-12-19 16:26 GMT+01:00 Yves :

Hi,

I am pulling my hair for days now...

I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register with 
my Asterisk.

There are no SIP Packets arriving at my asterisk at all... and it has nothing 
to do with a firewall or similar...

Simple Question:
Does anybody have a running SoundStation IP 6000 registerd with asterisk?

 

yes, I've got several of them running.
 

If so... would you please be so kind to tell me whats wrong with my setup?

AsteriskServer: 192.168.1.211
SIP-user: 165

(the SIP-Settings on asterisk-side are OK, tested with a normal Softphone... 
registering and placing calls is no problem...)

The phone-log only says: "Registration failed User: 165, Error Code:480 
Temporarily not available"

I tried with newest firmware, resetting to factory 100 times, using a 
provisionig file (which the SoundStation correctly downloads)
but it is always the same... the SoundStation does not contact the asterisk for 
registering...


1. Do you have any switch able to mirror traffic sent and received by Polycom 
phone ?

Capturing such traffic would help to understand what's happening.

2. Some phones support zero touch config with which they download their config 
files from the Internet.

Are you sure this doesn't happen ?

3. Is SNTP/NTP correctly configured on the phone ?



 

 


Phoneversion:


Telefoninformationen 


Telefonmodell 

SoundStation IP 6000 


Teilenummer 

3111-15600-001 Rev:W 


MAC-Adresse 

00:04:F2:07:0C:D3 


IP-Adresse 

192.168.0.13 


UC-Softwareversion 

4.0.11.0583 


BootROM-Softwareversion 

5.0.5.2324 


I can ping the phone from the asterisk, the phone can reach the asterisk server 
(as it downloads the tftp files, if used with
a provisioning profile), so the route and everything is correct... I even 
connected another Hardphone on the same cable
that stuck in the Polycom... no problem... the other phone can register and 
works, so there is really no cable or firewall
related problem here... it must be a setting!

thank you





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[asterisk-users] T1 -Asterisk server - Analog lines

[Motty Cruz]

Hello All, 

The problem I'm facing is the following: two machines that require analog
signal. However, connection to the world is setup as follow: T1 connection
to Asterisk server

Any suggestions on how to convert digital signal to analog? 

Type of card on my Asterisk server is wct4xxp

 

Thanks for your support!

Motty

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Re: [asterisk-users] Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers

[Motty Cruz]

It does not append the area code: 
5007-0047","SIP/truck1-0048","Dial","SIP/truck1/6052736,80","2016-11
-06 18:49:41",,"2016-11-06 18:49:45",4,0,"NO
ANSWER","DOCUMENTATION","1478458181.738",""
5007-0049","SIP/truck1-004a","Dial","SIP/truck1/6052736,80","2016-11
-06 18:52:26",,"2016-11-06 18:52:34",7,0,"NO
ANSWER","DOCUMENTATION","1478458346.826",""
5007-004b","SIP/truck1-004c","Dial","SIP/truck1/6052736,80","2016-11
-06 18:54:02",,"2016-11-06 18:54:23",20,0,"NO
ANSWER","DOCUMENTATION","1478458442.914",""

Our Sip provider takes 10 digits, it should have append 381 to 6052736
number. 

Thanks, 
Mottyh

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Sunday, November 06, 2016 12:15 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 to Asterisk 13.11 appending area
code to local numbers

On Sun, Nov 6, 2016, at 03:35 PM, Motty Cruz wrote:
> Hello, I would like to add area code to local numbers, it worked like 
> a charm on Asterisk 1.8 but does not work on Asterisk 13.11.
> 
>  
> 
> Extensions.conf; worked before on Asterisk 1.8 ; Adding Area code to 
> local numbers
> 
> exten => _9XXX,n,Set(CALLERID(all)="$CallerID" <3818008000>)
> 
> exten => _9XXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80)

This should also work fine. You'll need to provide the console output of an
attempt. Different logic may be executed.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers

[Motty Cruz]

Hello, I would like to add area code to local numbers, it worked like a
charm on Asterisk 1.8 but does not work on Asterisk 13.11. 

 

Extensions.conf; worked before on Asterisk 1.8
; Adding Area code to local numbers

exten => _9XXX,n,Set(CALLERID(all)="$CallerID" <3818008000>)

exten => _9XXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80)

 

Any ideas? 

Thanks, 
Motty

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Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

[motty cruz]

Thank you for your help! Centos 7 firewall was enable.

systemctl stop firewalld

issue fixed.

Thanks,


On Thu, Oct 13, 2016 at 3:54 PM, Victor Villarreal 
wrote:

> Ok.
>
> Please, note that 192.168.1.37 (I suspect) is the internal  LAN address Of
> the Polycom hardphone. If this is true, then you have  NAT issues.
>
> The REGISTER message are received by your PBX, but when respond, Asterisk
> send the next SIP message to the IP informed by the phone, that is the
> internal LAN address. The messages do not reach back to the hardphone.
>
> You need to setup a STUN server in the Polycom hardphone settings. Please,
> check the manual. Search in Google some public  STUN server to put in the
> settings.
>
> Last, the idea behind the "sip set debug" command was view the complete
> SIP messages conversation, not search for an error.
>
> On NAT escenarios, remember:
>
> * The NATed phones need to know the public  IP of the NATing router.
> Either by manual setting  or  by STUN protocol.
>
> * Reduce the time between REGISTERs attempt, if the client  have a dynamic
> IP connection.
>
> * Use the "localnet" SIP settings in Asterisk, so the PBX can distingish
> what Network need contacted via NAT and what not.
>
> Cheers.
>
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>   http://www.asterisk.org/community/astricon-user-conference
>
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>
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-- 
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Motty
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Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

[Motty Cruz]

Hello Victor, 

I did set debug on, but I don’t see any errors. I did tcpdump, client is trying 
to register:  here is the header of a udp packet

 

User Datagram Protocol, Src Port: 55300, Dst Port: 5060

Session Initiation Protocol (REGISTER)

Request-Line: REGISTER sip:pbx.mydomain.com:5060 SIP/2.0

Method: REGISTER

Request-URI: sip:pbx.mydomain.com:5060

[Resent Packet: True]

[Suspected resend of frame: 14]

Message Header

Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK7b2855394DB988BE

Transport: UDP

Sent-by Address: 192.168.1.37

Sent-by port: 5060

Branch: z9hG4bK7b2855394DB988BE

From: "1006" <sip:1...@pbx.mydomain.com>;tag=2859342B-CBC71460

SIP Display info: "1006"

SIP from address: sip:1...@pbx.mydomain.com

SIP from tag: 2859342B-CBC71460

To: <sip:1...@pbx.mydomain.com>

SIP to address: sip:1...@pbx.mydomain.com

SIP to address User Part: 1006

SIP to address Host Part: pbx.mydomain.com

CSeq: 1 REGISTER

Call-ID: 6cbe37bb-cca69d70-85d0431d@192.168.1.37

Contact: <sip:1006@192.168.1.37:5060>;methods="INVITE, ACK, BYE, 
CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.10.0689

Accept-Language: en

Max-Forwards: 70

Expires: 90

Content-Length: 0

 

Sip.conf 

[1006]

type=friend

username=1006

secret=mysecret

context=sip-phone

call-limit=5

callerid="iuser" <1006>

disallow=all

host=dynamic

allow=all

nat=yes

 

Is NAT value set to yes OK? Servers is on public IP, client is on private 
network. 

 

Thanks, 
Motty

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor Villarreal
Sent: Thursday, October 13, 2016 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 
64bit

 

Hi Motty,

Please, set  Verbose  to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug 
on".

Now try to register again. At last, " sip  de debug off".

Examine tour console  or  full log file to find some clue ir send me back some 
trace.

Cheers.

 

El oct. 13, 2016 1:45 PM, "Motty Cruz" <motty.c...@gmail.com> escribió:

Hello, fresh install of Asterisk 13.11.2, client unable to register.  For now I 
have IPtables disabled, also selinux is disabled

 

[1006]

type=friend

username=1006

secret=mysecret

context=sip-phone

call-limit=1

callerid="iuser" <1006>

disallow=all

host=dynamic

allow=all

 

any ideas? 

 

Thanks, 

Motty


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[asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

[Motty Cruz]

Hello, fresh install of Asterisk 13.11.2, client unable to register.  For
now I have IPtables disabled, also selinux is disabled

 

[1006]

type=friend

username=1006

secret=mysecret

context=sip-phone

call-limit=1

callerid="iuser" <1006>

disallow=all

host=dynamic

allow=all

 

any ideas? 

 

Thanks, 

Motty

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[asterisk-users] Asterisk Secure SIP session TLS port 5061

[Motty Cruz]

I finally secure SIP session between Asterisk server and a remote client. My
questions is the following; do I need to open port 5061 UDP on my firewall
or just port 5061 TCP for SIP sessions.? 
I am not interested in securing RTP only SIP sessions. 

 

Thanks for your help!

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Re: [asterisk-users] Asterisk 1.8 secure SIP session only

[Motty Cruz]

Thank you Markos, finally was able to secure SIP session with TLS between 
server & client. 

 

Thanks for you support!

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markos Vakondios
Sent: Wednesday, May 04, 2016 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8 secure SIP session only

 

Your CA cert is missing.

 

Add in sip.conf:

 

tlscafile=/etc/asterisk/keys/ca.crt

 

You don't need:

tlscapath=/etc/asterisk/keys

 

On 4 May 2016 at 19:43, Motty Cruz <motty.c...@gmail.com> wrote:

Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I 
keep getter an error, 

  == Problem setting up ssl connection: error:14094418:SSL 
routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
[2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: 
FILE * open failed!

I tried both signed and self-signed cert to no avail. 

Here is my Configuration: 

Sip.conf

tlsenable=yes

tlsbindaddr=0.0.0.0

tlscertfile=/etc/asterisk/keys/box1.pem

tlscapath=/etc/asterisk/keys

tlscipher=ALL

tlsclientmethod=tlsv1

 

sip.conf ext.

[5006]

type=peer

context=sipext

call-limit=3

trustrpid=no

callerid="Rec" <5006>

disallow=all

allow=ulaw

allow=alaw

username=5006

secret=9fcbb025200881850526bc57d59885c3

dtmfmode=rfc2833

host=dynamic

mailbox=5006

nat=yes

canreinvite=no

transport=tls

 

  == Problem setting up ssl connection: error:14094418:SSL 
routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
[2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: 
FILE * open failed!

Any ideas? 

 


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[asterisk-users] Asterisk 1.8 secure SIP session only

[Motty Cruz]

Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I
keep getter an error, 

  == Problem setting up ssl connection: error:14094418:SSL
routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
[2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection:
FILE * open failed!

I tried both signed and self-signed cert to no avail. 

Here is my Configuration: 

Sip.conf

tlsenable=yes

tlsbindaddr=0.0.0.0

tlscertfile=/etc/asterisk/keys/box1.pem

tlscapath=/etc/asterisk/keys

tlscipher=ALL

tlsclientmethod=tlsv1

 

sip.conf ext.

[5006]

type=peer

context=sipext

call-limit=3

trustrpid=no

callerid="Rec" <5006>

disallow=all

allow=ulaw

allow=alaw

username=5006

secret=9fcbb025200881850526bc57d59885c3

dtmfmode=rfc2833

host=dynamic

mailbox=5006

nat=yes

canreinvite=no

transport=tls

 

  == Problem setting up ssl connection: error:14094418:SSL
routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
[2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection:
FILE * open failed!

Any ideas? 

 

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[asterisk-users] dahdi auto-call multiple destinations

[motty cruz]

Am trying to get a script to call multiple destinations, for instance I
have the following file:

Channel: dahdi/g1/6078880
CallerID: "Room Tempeture" <800579>
MaxRetries: 2
RetryTime: 60
WaitTime: 20

This works great for me, however I am trying to add a secondary number, is
that possible to add dahdi/g1/6078880/g1/7068880 ?

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Motty
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[asterisk-users] Asterisk 1.8.22.0 built - encrypt authentication

[Motty Cruz]

Hello,
I would like to encrypt password between Asterisk servers and clients. 
is there an easy way to do so? I am running Asterisk 1.8.22.0 built on 
CentOS 6.3


Thanks,
.Motty
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Re: [asterisk-users] Asterisk 1.8.22.0 built - encrypt authentication

[Motty Cruz]

Thanks Oliver,
I am using this: http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret

however, I wanted to know if there is a way to use some sort of 
Certificates.

http://www.voip-info.org/wiki/view/Asterisk+encryption

has anybody used this feature?

Thanks,
.Motty

On 07/29/2015 03:40 PM, Oli-net wrote:

Hello,

Try MD5SUM like that in a terminal
echo -n myverysecretpassword | md5sum
It will return you something like that 
22ea6cf875d66b15d275684427275dfdf witch is your password in an MD5 format.


Hope this help
Oliver

Le 29/07/2015 16:30, Motty Cruz a écrit :

Hello,
I would like to encrypt password between Asterisk servers and 
clients. is there an easy way to do so? I am running Asterisk 
1.8.22.0 built on CentOS 6.3


Thanks,
.Motty








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Re: [asterisk-users] Sending E-Mail from voicemail with AND without attachment

[Motty Cruz]

Hello,
this worked for me:
500 = mysecret,Motty 
mailbox,mo...@domain.com,,tz=pacific,attach=yes|delete=1


Thanks,
Motty

On 07/10/2015 07:45 AM, Luca Bertoncello wrote:

Hi again,

I'm trying to send two E-Mails when a message comes in the voicemail, 
the first WITH the attachment, and the second WITHOUT.

But I don't get it working...

I wrote

004935 = SECRET,John Doe,first@emailde,sec...@email.de,attach=no

but both E-Mail don't have the attachment.
Writing

004935 = SECRET,John Doe,first@emailde,attach=no|sec...@email.de

results in invalid E-Mail-Address for the second E-Mail.

I really don't know how to solve my problem, other the write a little 
program on the mail server that remove the attachment from the second 
E-Mail...


Any help is appreciated
Luca Bertoncello
(lucab...@lucabert.de)





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Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

[Motty Cruz]

Here is what i have,
exten = _011X,1,Set(CALLERID(number)=factory2)
exten = _011X,2,Authenticate(/home/asterisk/passwds.conf,m,3)
exten = _011X,3,NoOp(user has been authenticated)
exten = _011X,n,Dial(SIP/VoIPSP1/${EXTEN:1},80)
exten = _011X,n,HangUp()

I would like to add background music if authentication failed, then 
after 6 minutes hangup


any ideas, suggestions?

On 07/07/2015 09:09 AM, Motty Cruz wrote:

Hello,
I used this guide, it worked for me:
http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html

Thanks,

On 07/06/2015 04:54 PM, John Kiniston wrote:

The Authenticate application will do this for you.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate

You can either give it a single PIN to use for all calls, 
Authenticate using a value in the Asterisk Database, Or use a plain 
text file for the PIN's





On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz motty.c...@gmail.com 
mailto:motty.c...@gmail.com wrote:


Hello All,

I will like to configure Asterisk to use PIN Code for all
outgoing international calls.

Also, any suggestions as to when should I prompt users for code
prior to dialing the number or after dialing the number?

can someone provide with a example on how to accomplish this
goal? I am a bit confuse by this :

http://forums.digium.com/viewtopic.php?p=130936sid=707f657f7a61dfed55e4922304925091

Thanks for your help.


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Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

[Motty Cruz]

Hello,
I used this guide, it worked for me:
http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html

Thanks,

On 07/06/2015 04:54 PM, John Kiniston wrote:

The Authenticate application will do this for you.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate

You can either give it a single PIN to use for all calls, Authenticate 
using a value in the Asterisk Database, Or use a plain text file for 
the PIN's





On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz motty.c...@gmail.com 
mailto:motty.c...@gmail.com wrote:


Hello All,

I will like to configure Asterisk to use PIN Code for all outgoing
international calls.

Also, any suggestions as to when should I prompt users for code
prior to dialing the number or after dialing the number?

can someone provide with a example on how to accomplish this goal?
I am a bit confuse by this :

http://forums.digium.com/viewtopic.php?p=130936sid=707f657f7a61dfed55e4922304925091

Thanks for your help.


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butcher a hog, conn a ship, design a building, write a sonnet, balance 
accounts, build a wall, set a bone, comfort the dying, take orders, 
give orders, cooperate, act alone, solve equations, analyze a new 
problem, pitch manure, program a computer, cook a tasty meal, fight 
efficiently, die gallantly. Specialization is for insects.

---Heinlein




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[asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

[Motty Cruz]

Hello All,

I will like to configure Asterisk to use PIN Code for all outgoing 
international calls.


Also, any suggestions as to when should I prompt users for code prior to 
dialing the number or after dialing the number?


can someone provide with a example on how to accomplish this goal? I am 
a bit confuse by this : 
http://forums.digium.com/viewtopic.php?p=130936sid=707f657f7a61dfed55e4922304925091


Thanks for your help.


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[asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user

[Motty Cruz]

Hello,
I would like to setup a mechanism to trigger an alarm if user is deal 
too many numbers within a very short period of time. Safeguard against 
users hacked accounts.


can someone help?

Thanks,

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Re: [asterisk-users] adding area code

[Motty Cruz]

this code worked for me,

here is what I did and worked for me:

exten = 1381+NXX,1,Set(CALLERID(number)=3817383444)

exten = 1+NXXNXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)


Thanks for you help!

On 04/27/2015 02:56 PM, Matt Riddell wrote:


On 27Apr, 2015, at 16:39, Motty Cruz motty.c...@gmail.com 
mailto:motty.c...@gmail.com wrote:


forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS.

Thanks,


On 04/27/2015 02:38 PM, Motty Cruz wrote:

here is what I have:

exten = _9XXX,1,Set(l_HomeAreaCode=381)

exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1})

exten = _9XXX,n,Dial(SIP/SIP-Provider/${dialnumber},80)

not having success;

Got SIP reponse 503 Service Unavailable”


Can you send us the console output when you make the call?

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Re: [asterisk-users] adding area code

[Motty Cruz]

I apologize, I coppied the wrong code,
here is the code I am using:

; Adding Area code and striping 9 for local numbers
exten = _9XXX,n,Set(CALLERID(all)= 3817383444)
exten = _9XXX,n,Dial(SIP/intelepeer/1381${EXTEN:1},80)


Thanks,
motty

On 04/28/2015 11:54 AM, Chad Wallace wrote:

On Tue, 28 Apr 2015 07:21:12 -0700
Motty Cruz motty.c...@gmail.com wrote:


here is what I did and worked for me:

exten = 1381+NXX,1,Set(CALLERID(number)=3817383444)

exten = 1+NXXNXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)

I find it hard to believe this is working.

First, you don't have a leading underscore on your patterns.  Your
users aren't literally dialing the N's and X's are they?

Second, what's with the plus in the extension?  You want your users to
dial that?

Third, that's two different extensions, one with priority 1 and one
with priority 2.  The first one will set a variable and hangup, and the
second there's no priority 1 for that extension... I've never tried
that... I'm assuming it just won't work.




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Re: [asterisk-users] adding area code

[Motty Cruz]

Thanks for your reply,

[globals]
AREACODE=381

[outbound]
exten = _NXX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN},80)

did not work for me, any ideas?

Thanks,

On 04/27/2015 01:59 PM, Phil Reynolds wrote:



On 27 April 2015 21:32:42 BST, Motty Cruz motty.c...@gmail.com wrote:
Hello,

I would like to add area code if clients dial 7 digits, it that
possible? currently clients dial prefix 9 plus local number, however my

SIP provider is requiring to dial 10 digits. is it possible to add area

code?r

Quite simple - you need to match on NXXX and when passing it to 
the SIP provider, present ${AREACODE}${EXTEN}, having first defined 
AREACODE in [globals].


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Re: [asterisk-users] adding area code

[Motty Cruz]

here is what I have:

exten = _9XXX,1,Set(l_HomeAreaCode=381)

exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1})

exten = _9XXX,n,Dial(SIP/SIP-Provider/${dialnumber},80)

not having success;

Got SIP reponse 503 Service Unavailable

On 04/27/2015 02:19 PM, Bryant Zimmerman wrote:

Motty
Yes
From your dial plan accept 9 + 7 digits then concat your dialed number 
together with your areacode.

This s a brief example.
exten = _9XXX,1,Set(l_HomeAreaCode=555)
exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; 
This line should combine your area code and the last 7 digits of your 
dialed phone number

exten = _9XXX,n,Dial(SIP/${dialnumber},35)
Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

*From*: Motty Cruz motty.c...@gmail.com
*Sent*: Monday, April 27, 2015 4:33 PM
*To*: asterisk-users@lists.digium.com
*Subject*: [asterisk-users] adding area code
Hello,

I would like to add area code if clients dial 7 digits, it that
possible? currently clients dial prefix 9 plus local number, however my
SIP provider is requiring to dial 10 digits. is it possible to add area
code?

Thanks,
Motty

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[asterisk-users] adding area code

[Motty Cruz]

Hello,

I would like to add area code if clients dial 7 digits, it that 
possible? currently clients dial prefix 9 plus local number, however my 
SIP provider is requiring to dial 10 digits. is it possible to add area 
code?


Thanks,
Motty

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Re: [asterisk-users] adding area code

[Motty Cruz]

forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS.

Thanks,


On 04/27/2015 02:38 PM, Motty Cruz wrote:

here is what I have:

exten = _9XXX,1,Set(l_HomeAreaCode=381)

exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1})

exten = _9XXX,n,Dial(SIP/SIP-Provider/${dialnumber},80)

not having success;

Got SIP reponse 503 Service Unavailable

On 04/27/2015 02:19 PM, Bryant Zimmerman wrote:

Motty
Yes
From your dial plan accept 9 + 7 digits then concat your dialed 
number together with your areacode.

This s a brief example.
exten = _9XXX,1,Set(l_HomeAreaCode=555)
exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; 
This line should combine your area code and the last 7 digits of your 
dialed phone number

exten = _9XXX,n,Dial(SIP/${dialnumber},35)
Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

*From*: Motty Cruz motty.c...@gmail.com
*Sent*: Monday, April 27, 2015 4:33 PM
*To*: asterisk-users@lists.digium.com
*Subject*: [asterisk-users] adding area code
Hello,

I would like to add area code if clients dial 7 digits, it that
possible? currently clients dial prefix 9 plus local number, however my
SIP provider is requiring to dial 10 digits. is it possible to add area
code?

Thanks,
Motty

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[asterisk-users] how to strip +1 out of incoming number

[motty cruz]

Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?

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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

[motty cruz]

Thank you AJ, I will certainly not copy and past; I want to believe I
understand the risk. I needed some kind of direction, thank you for your
support.

-Motty

On Fri, Sep 19, 2014 at 2:51 AM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:

 On Thursday 18 Sep 2014, motty cruz wrote:
  Hello, I would to allow users to place calls overseas such as India and
  Malaysia but only with a security code. if they don't have a security
 code
  I want to be able to drop the calls.
 
  can someone point me to a right direction to achieve this goal?
 
  Thanks,
  Motty

 Not many people are going to want to answer this definitively, I suspect,
 for
 fear of being blamed if you copy what they did, it doesn't work for you and
 you get landed with huge bills for calls you didn't make.  Securing
 Asterisk
 is never as easy as you think.


 However, if you look back through my own posts, I did post some dialplan
 code
 a short while ago, relating to a PIN entry.  Feel free to borrow that and
 play
 around with it; but note, I will not accept any responsibility for it not
 being as secure as you thought!


 Another thing to consider would be only allowing overseas calls from a
 particulat context; any extension that does not require the ability to call
 abroad should be placed in a different default context.  If you know you
 will
 only ever need to call a restricted range of foreign numbers, consider
 giving
 them short codes -- endpoints effectively within your own internal
 numbering
 scheme -- and sending calls to _00X. to a recorded message.

 [overseas-offices]
 ; this context is only for phones which need the ability to call overseas

 ; 8000 is office in France
 exten = 8000,1,Set(CALLERID(num)=${OUTGOING_IDENT})
 exten = 8000,n,Dial(${OUT_TRUNK}/0033251478820,180)
 exten = 8000,n,Hangup()

 ; 8010 is office in India
 exten = 8010,1,Set(CALLERID(num)=${OUTGOING_IDENT})
 exten = 8010,n,Dial(${OUT_TRUNK}/00918322494200,180)
 exten = 8010,n,Hangup()

 ; .

 [default]
 ; play suitably sarchastic announcement to chancers
 _00X.,1,Play(ajs-not_allowed)
 _00X.,n,Hangup()


 Basically, be paranoid; and even then, don't forget, you probably aren't
 being
 paranoid enough.

 --
 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] Asterisk prefix code to dial a high fraud country - security mechanism

[motty cruz]

Hello, I would to allow users to place calls overseas such as India and
Malaysia but only with a security code. if they don't have a security code
I want to be able to drop the calls.

can someone point me to a right direction to achieve this goal?

Thanks,
Motty
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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

[motty cruz]

Thank you Julian,

would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.

Thanks,
Motty

On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk wrote:

 Hello motty,

 Thursday, September 18, 2014, 6:35:40 PM, you wrote:

  Hello, I would to allow users to place calls overseas such as India
  and Malaysia but only with a security code. if they don't have a
  security code I want to be able to drop the calls.

 I use this

 exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
 same = n,Playback(silence/1)
 same = n,Authenticate(9084,,4)
 same = n,Macro(outgoingTrunk,${EXTEN})
 same = n,Hangup()

 It  uses  a  fixed PIN number which calls a macro which deals with the
 actual  dialling,  but  a  standard  Dial command would work here too.
 Quick  and  easy, but there are lots of options. If the correct PIN is
 not entered, the call is not made.

 --
 Best regards,
  Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

[motty cruz]

Thanks Eric, for respectfully pointing that link, it is the reason why I am
posting my question for lack of knowledge. I had been working on Asterisk
for the last 4 years, I am always learning something knew.

- Motty

On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling ewiel...@nyigc.com wrote:

 Your question demonstrates a fundamental lack of Asterisk concepts and
 knowledge.  You should start by reading http://www.asteriskdocs.org/ and
 go from there.Asterisk is not something you can learn in a few days.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz
 *Sent:* Thursday, September 18, 2014 4:52 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud
 country - security mechanism



 Thank you Julian,



 would it be possible to block calls to international calls except certain
 countries? I just want to make sure that if attackers try to place calls
 outside the states they not succeed.



 Thanks,
 Motty



 On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk
 wrote:

 Hello motty,

 Thursday, September 18, 2014, 6:35:40 PM, you wrote:

  Hello, I would to allow users to place calls overseas such as India
  and Malaysia but only with a security code. if they don't have a
  security code I want to be able to drop the calls.

 I use this

 exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
 same = n,Playback(silence/1)
 same = n,Authenticate(9084,,4)
 same = n,Macro(outgoingTrunk,${EXTEN})
 same = n,Hangup()

 It  uses  a  fixed PIN number which calls a macro which deals with the
 actual  dialling,  but  a  standard  Dial command would work here too.
 Quick  and  easy, but there are lots of options. If the correct PIN is
 not entered, the call is not made.

 --
 Best regards,
  Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

[motty cruz]

absolutely not what I meant, I really meant to say thank you for
respectfully pointing that out.


-Motty

On Thu, Sep 18, 2014 at 2:32 PM, Eric Wieling ewiel...@nyigc.com wrote:

 It is unfortunate
 http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-3.6
 is not helpful to you.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz
 *Sent:* Thursday, September 18, 2014 5:27 PM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud
 country - security mechanism



 Thanks Eric, for respectfully pointing that link, it is the reason why I
 am posting my question for lack of knowledge. I had been working on
 Asterisk for the last 4 years, I am always learning something knew.



 - Motty



 On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling ewiel...@nyigc.com wrote:

 Your question demonstrates a fundamental lack of Asterisk concepts and
 knowledge.  You should start by reading http://www.asteriskdocs.org/ and
 go from there.Asterisk is not something you can learn in a few days.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz
 *Sent:* Thursday, September 18, 2014 4:52 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud
 country - security mechanism



 Thank you Julian,



 would it be possible to block calls to international calls except certain
 countries? I just want to make sure that if attackers try to place calls
 outside the states they not succeed.



 Thanks,
 Motty



 On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk
 wrote:

 Hello motty,

 Thursday, September 18, 2014, 6:35:40 PM, you wrote:

  Hello, I would to allow users to place calls overseas such as India
  and Malaysia but only with a security code. if they don't have a
  security code I want to be able to drop the calls.

 I use this

 exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
 same = n,Playback(silence/1)
 same = n,Authenticate(9084,,4)
 same = n,Macro(outgoingTrunk,${EXTEN})
 same = n,Hangup()

 It  uses  a  fixed PIN number which calls a macro which deals with the
 actual  dialling,  but  a  standard  Dial command would work here too.
 Quick  and  easy, but there are lots of options. If the correct PIN is
 not entered, the call is not made.

 --
 Best regards,
  Julianmailto:jb_s...@trink.co.uk


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[asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf

[motty cruz]

Hello,
a user outside the office regularly gets a call from ext. 101 but that
extension does not exist in my extensions.conf. when the user pickup the
phone no one answers. Any Idea how to fix this issue? that user uses
Polycom SP 450,

Thanks in advance,
Motty
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Re: [asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf

[motty cruz]

Thanks Eric, for point to polycom instructions I will give it a try.

Kevin, I am sure called is not originating from our system,

Thanks for your support.

On Tue, Sep 16, 2014 at 9:08 AM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:

  Hello,
  a user outside the office regularly gets a call from ext. 101 but
  that extension does not exist in my extensions.conf. when the user
  pickup the phone no one answers. Any Idea how to fix this issue?
  that user uses Polycom SP 450,

 First thing to look at is at the time the user receives the call, do you
 show anything in your Asterisk CLI? I would make sure that the call is
 actually originating from your system and track back from there.
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Re: [asterisk-users] chan_sip.c:23647 handle_request_invite: Failed to authenticate device

[motty cruz]

Hello Deepak,

601sip:601@111.118.185.107;tag=2f498fbd

is the 111.118.185.107 your server IP? it could be a client trying to
authenticate as Rusty suggest or an attacker attempting to gain access, if
you want to find out what IP address that request is coming from do the
following command. make sure you have tcpdump installed.

tcpdump -lni eth0 -f udp port 5060

monitor your server to make sure you catch the attacker, also you could do

tcpdump -w capture.cap

and analyze it with wireshark.

Thanks,
Motty

On Thu, Sep 11, 2014 at 2:28 PM, Rusty Newton rnew...@digium.com wrote:

 On Thu, Sep 11, 2014 at 10:14 AM, Deepak Bhatia dee...@voxomos.com
 wrote:
  Hi,
 
  Why are we getting message in the asterisk
 
  [Sep 10 12:55:23] NOTICE[15043]: chan_sip.c:23647 handle_request_invite:
  Failed to authenticate device 601sip:601@111.118.185.107;
  tag=2f498fbd
  [Sep 10 12:55:24] NOTICE[15043]: chan_sip.c:23647 handle_request_invite:
  Failed to authenticate device 601sip:601@111.118.185.107;tag=209a8aa9
 
  Regards
 
  Deepak Bhatia

 Deepak,

  This is commonly a wrong password on the client-side. That is, the
 device attempting to call Asterisk is failing an authentication
 challenge for one reason or another.

  For anyone to help you, you would probably need to post your sip.conf
 configuration (full) and screenshots of your soft/hard client
 configuration.

  Obviously you would want to use a fake password and sanitize IP
 addresses if necessary.

 Thanks,

 --
 Rusty Newton
 Digium, Inc. | Community Support Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 direct: +1 256 428 6200

 Check us out at: http://digium.com  http://asterisk.org

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[asterisk-users] Asterisk failed to authenticate device - attack attempt.

[motty cruz]

Hi all,
I continue to see the following msg on my Asterisk log:

[Sep  8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite:
Failed to authenticate device 9009sip:9...@196.107.xx.xx;tag=8dd48dd2

IP: 196.107.xx.xx is my asterisk server IP address.

I don't know what it means and how to cover any holes that attacker is
trying to exploit.

Thanks,
Motty
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[asterisk-users] Asterisk secure fine tune - stop attack

[motty cruz]

Hi All,
I see this kind of attack on our Asterisk Server, do you know how to block
that IP?

[Sep  4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite:
Call from '' (213.136.81.166:9306) to extension '34422' rejected because
extension not found in context 'default'.

Thanks in advance,
-Motty
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Re: [asterisk-users] Asterisk secure fine tune - stop attack

[motty cruz]

Thanks, looks like fail2ban is the way to go, I would prefer a different
alternatives if there is one. I tried deny=IP/netmask but did not work for
me, in sip.conf. seems like fail2ban is what you all are using, so I will
give it a try.

Thanks,


On Thu, Sep 4, 2014 at 7:58 AM, Thorsten Göllner t...@ovm-group.com wrote:


 Am 04.09.2014 16:44, schrieb motty cruz:

  Hi All,
 I see this kind of attack on our Asterisk Server, do you know how to block
 that IP?

  [Sep  4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite:
 Call from '' (213.136.81.166:9306) to extension '34422' rejected because
 extension not found in context 'default'.


 You should not invest time in blocking single IPs. Take a look at
 fail2ban.
 http://www.fail2ban.org/wiki/index.php/Asterisk

 -Thorsten-

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Re: [asterisk-users] Asterisk secure fine tune - stop attack

[motty cruz]

Hi A J,
believe me, I wish i do as you suggested, however I have a few extensions
outside the office with dynamic IPs, so that is not a possibility. Thanks
for your suggestions, I will try fail2ban. I don't know how complicated is
to implement that on production server.

Thanks,
-Motty


On Thu, Sep 4, 2014 at 8:19 AM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:

 On Thursday 04 Sep 2014, motty cruz wrote:
  Hi All,
  I see this kind of attack on our Asterisk Server, do you know how to
 block
  that IP?

 Instead of blocking unwanted IPs, you should be permitting only wanted IPs.

 --
 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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Re: [asterisk-users] Asterisk secure fine tune - stop attack

[motty cruz]

Thank you all for your support, your suggestions are welcome.
Thanks,


On Thu, Sep 4, 2014 at 9:26 AM, Chris Bagnall aster...@lists.minotaur.cc
wrote:

 On 4/9/14 4:58 pm, Eric Wieling wrote:

 If we don't need to allow access from outside the USA we block access
 from all non-ARIN IP addresses by using iptables.   This takes care of at
 least 80% of attacks.


 Likewise here (though RIPE rather than ARIN, since we're the other side of
 the pond).

 You can also take it a bit further: if, for example, you know what ISP(s)
 your dynamic clients are using, you can limit connections to the IP ranges
 those ISP(s) use - look up their ranges on he.net's BGP looking glass if
 you need to find out what ranges they're using.

 Another thing I've been playing with of late is using iptables' string
 matching functionality to block user agents of known attack vectors:
 'sipcli', 'sipvicious', 'friendly-scanner', etc.

 This seems to work remarkably well, though what impact it has on net
 performance under load remains to be seen.

 Kind regards,

 Chris
 --
 This email is made from 100% recycled electrons


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[asterisk-users] share mailbox Asterisk 1.8.22

[motty cruz]

Hello, I want to share mailbox between two extensions
Ext. 101
Ext. 102

I want the messages to go to mailbox 101, when when checked mailbox from
extension 102 to be able to clear the bliking red light.

here is extensions.conf
exten = 102,hint,SIP/${EXTEN}
exten = 102,1,Dial(SIP/101SIP/102,20,t)
exten = 102,2,Voicemail(101,u)
exten = 102,102,Voicemail(101,b)
exten = 102,103,Hangup

sip.conf
[102]
type=friend
context=sipphones
call-limit=99
callerid=Jo 102
disallow=all
allow=ulaw
allow=alaw
username=102
secret=xexpasswd
dtmfmode=rfc2833
host=dynamic
mailbox=101
nat=yes
canreinvite=no

this configuration work ok but, light msg keeps bilking after checking for
mesages.

Any suggestions?
Thanks,
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Re: [asterisk-users] share mailbox Asterisk 1.8.22

[motty cruz]

yes they're able to hear the same msg, in /var/spool/asterisk/default/
rm -rf 102
ln -s 101 102

but it does not clear out,

Thanks,


On Tue, Jun 24, 2014 at 3:31 PM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:

 I have done this for one of my users in a very similar fashion. When 102
 checks the voicemail, do they hear the correct voicemails? Ours clears just
 fine in this situation.

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208

 asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 04:37:26 PM:

  From: motty cruz motty.c...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com,
  Date: 06/24/2014 04:37 PM
  Subject: [asterisk-users] share mailbox Asterisk 1.8.22
  Sent by: asterisk-users-boun...@lists.digium.com
 
  Hello, I want to share mailbox between two extensions
  Ext. 101
  Ext. 102
 
  I want the messages to go to mailbox 101, when when checked mailbox
  from extension 102 to be able to clear the bliking red light.
 
  here is extensions.conf
  exten = 102,hint,SIP/${EXTEN}
  exten = 102,1,Dial(SIP/101SIP/102,20,t)
  exten = 102,2,Voicemail(101,u)
  exten = 102,102,Voicemail(101,b)
  exten = 102,103,Hangup
 
  sip.conf
  [102]
  type=friend
  context=sipphones
  call-limit=99
  callerid=Jo 102
  disallow=all
  allow=ulaw
  allow=alaw
  username=102
  secret=xexpasswd
  dtmfmode=rfc2833
  host=dynamic
  mailbox=101
  nat=yes
  canreinvite=no
 
  this configuration work ok but, light msg keeps bilking after
  checking for mesages.
 
  Any suggestions?
  Thanks,
 
 
  __
  This email has been scanned by the Symantec Email Security.cloud service.
  For more information please visit http://www.symanteccloud.com
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Re: [asterisk-users] share mailbox Asterisk 1.8.22

[motty cruz]

Thanks Kevin,

can you provide me with example of your code? if you don't mind?

Thanks,


On Tue, Jun 24, 2014 at 3:46 PM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:

 asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:36:16 PM:

  From: motty cruz motty.c...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com,
  Date: 06/24/2014 05:36 PM
  Subject: Re: [asterisk-users] share mailbox Asterisk 1.8.22
  Sent by: asterisk-users-boun...@lists.digium.com
 
  yes they're able to hear the same msg, in /var/spool/asterisk/default/

  rm -rf 102
  ln -s 101 102
 
  but it does not clear out,
 
  Thanks,
 

 That is where your setup and mine are different. I have my second
 extension directly check the first extensions mailbox as opposed to using a
 symlink. That way, when the box is cleared, it is actually happening in the
 original mailbox.

 Basically, my code to check voicemail uses the CALLERID(num) to determine
 the mailbox and I have both extensions set to use the same caller ID. This
 works for me as both extensions belong to the same person (one phone at the
 office, one at home) and they want to always have their outbound calls show
 up as if they were from their published number regardless of which phone
 they use.
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Re: [asterisk-users] Asterisk 1.8.22

[motty cruz]

Thanks for your support, I will try your suggestions,

will let you know how it goes,

Thanks,


On Tue, May 13, 2014 at 7:28 AM, Michelle Dupuis mdup...@ocg.ca wrote:

  Another alternative is SecAst (Asterisk intrusion detection system).
  Grab the free version from www.generationd.com​


  It does everything fail2ban does, plus you have the option of blocking
 IP's based on geograhic origin, detecting suspicious call patterns, etc.


  -=M=-

 All opinions posted are my own. But as an employee of GenerationD System
 my views are undoubtedly biased :)


  --
 *From:* asterisk-users-boun...@lists.digium.com 
 asterisk-users-boun...@lists.digium.com on behalf of motty cruz 
 motty.c...@gmail.com
 *Sent:* Monday, May 12, 2014 5:43 PM
 *To:* Asterisk Users List
 *Subject:* [asterisk-users] Asterisk 1.8.22

  Hello,
 recently I have seen spike in attacks on my asterisk server, this is what
 I get on the LCD of my phone: 201@76.220.5.205

 or calls from 1000 sip1000@76.2230.5.205,

  have any idea on how to stop this calls?

  Thanks,

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[asterisk-users] Asterisk 1.8.22

[motty cruz]

Hello,
recently I have seen spike in attacks on my asterisk server, this is what I
get on the LCD of my phone: 201@76.220.5.205

or calls from 1000 sip1000@76.2230.5.205,

have any idea on how to stop this calls?

Thanks,
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[asterisk-users] how to hangup Local/100 channel

[motty cruz]

Hello All,

one of the extensions fall into a loop, I don't know how to hangup that
channel

-- Executing [i@autoatten:2] Goto(Local/100@sipphones-01b2;2,
s,2) in new stack
-- Goto (autoatten,s,2)
-- Sent into invalid extension 's' in context 'autoatten' on
Local/200@sipphones-01b2;2
-- Executing [i@autoatten:1] Playback(Local/2000@sipphones-01b2;2,
pbx-invalid) in new stack
-- Local/200@sipphones-01b2;2 Playing 'pbx-invalid.gsm' (language
'en')
-- Executing [i@autoatten:2] Goto(Local/200@sipphones-01b2;2,
s,2) in new stack
-- Goto (autoatten,s,2)
-- Sent into invalid extension 's' in context 'autoatten' on
Local/200@sipphones-01b2;2
-- Executing [i@autoatten:1] Playback(Local/200@sipphones-01b2;2,
pbx-invalid) in new stack
-- Local/200@sipphones-01b2;2 Playing 'pbx-invalid.gsm' (language
'en')
-- Remote UNIX connection
-- Executing [i@autoatten:2] Goto(Local/200@sipphones-01b2;2,
s,2) in new stack
-- Goto (autoatten,s,2)
-- Sent into invalid extension 's' in context 'autoatten' on
Local/200@sipphones-01b2;2
-- Executing [i@autoatten:1] Playback(Local/200@sipphones-01b2;2,
pbx-invalid) in new stack
-- Local/200@sipphones-01b2;2 Playing 'pbx-invalid.gsm' (language
'en')
-- Executing [i@autoatten:2] Goto(Local/200@sipphones-01b2;2,
s,2) in new stack
-- Goto (autoatten,s,2)
-- Sent into invalid extension 's' in context 'autoatten' on
Local/200@sipphones-01b2;2
-- Executing [i@autoatten:1] Playback(Local/200@sipphones-01b2;2,
pbx-invalid) in new stack
-- Local/200@sipphones-01b2;2 Playing 'pbx-invalid.gsm' (language
'en')


any ideas?

Thanks,
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Re: [asterisk-users] how to hangup Local/100 channel

[motty cruz]

Thanks for your support, I was able to soft hangup using hangup request
Local/200@users-0001b

first, I did core show channels,
after stopping this loops I was able to fixed that problem from happening
again,

Thanks,


On Mon, May 5, 2014 at 8:56 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Mon, 5 May 2014, motty cruz wrote:

  one of the extensions fall into a loop, I don't know how to hangup that
 channel

 -- Goto (autoatten,s,2)
 -- Sent into invalid extension 's' in context 'autoatten' on
 Local/200@sipphones-01b2;2

 any ideas?


 If you're asking how to prevent it from happening, how about 'exten =
 s,2,hangup()?'

 If you're asking how to hang up the channel while it is in a loop, what
 have you tried? Does 'channel request hangup' help?

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
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Re: [asterisk-users] Asterisk 1.6

[motty cruz]

that is definitely another options, thanks for the range of options
provided,

Thanks


On Sat, Apr 5, 2014 at 4:51 PM, Duncan Turnbull dun...@e-simple.co.nzwrote:

 Another option we like, but i depends on your preferences is to run them
 over openvpn. Works for Mac, Linux and Windows clients.

 Since all out clients are under our control we use openvpn a lot and
 yealink and other phones have it built in so they can connect directly once
 initially setup

 Cheers Duncan

 On 5/04/2014, at 4:36 am, motty cruz motty.c...@gmail.com wrote:

 that sounds feasible, Thanks Michelle,




 On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote:

  If you know your users are all from with your country, or state, or
 even city, you could restrict geographic access in your secast.conf file
 like this:


 ruledefault=deny
  ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA

  The above would:
 - By default deny all source IP's anywhere in the world
 - Let in only source IP's from:
 1. North America (continent), Canada (country), Ontario (region)
 2. North America (continent), USA (country), Michigan (region), Detroit
 (city)
 3. Any region called 'Ohio' anywhere in the world (not sure why you would
 do that but fun example)
 4. Anywhere in North America

  So you can open up your system based solely on where you know your real
 users are located.

 -=Michelle=-


  --
 *From:* asterisk-users-boun...@lists.digium.com 
 asterisk-users-boun...@lists.digium.com on behalf of motty cruz 
 motty.c...@gmail.com
 *Sent:* Friday, April 4, 2014 11:15 AM

 *To:* Asterisk Users List
 *Subject:* Re: [asterisk-users] Asterisk 1.6

  Hello Ishfaq, outside users usually travel around the country and
 connect from different network, so it won't be possible to lock it down to
 specific IP.

  Thanks for your support.


 On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:




  On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote:

 thank you all for your support. I am using Linux, I only have about 7
 users outside our home network. I will learn fail2ban and will use it
 accordingly.

  again Thanks for your support.



Do the 7 users outside of your home network always connect from the
 same IP addresses? If so, you can just lock down your SIP port to those 7
 IPs explicitly in your IPTables configuration.

  Another option would be to change which port you're running SIP on.


  --

 Ishfaq Malik
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
 37 Ducie Street
 Manchester, M1 2JW
 COMPANY REG NO. 04920552


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[asterisk-users] Asterisk 1.6

[motty cruz]

Hello All, my asterisk server is constantly under attack

[Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password
[Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password
[Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password
[Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password
[Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password

is there a way to reject their registration after a three consecutive
tries?

Thanks,
Call
Send SMS
Add to Skype
You'll need Skype CreditFree via Skype
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Re: [asterisk-users] Asterisk 1.6

[motty cruz]

thank you all for your support. I am using Linux, I only have about 7 users
outside our home network. I will learn fail2ban and will use it
accordingly.

again Thanks for your support.


On Fri, Apr 4, 2014 at 7:09 AM, Mauricio Tavares raubvo...@gmail.comwrote:




 On Fri, Apr 4, 2014 at 10:05 AM, Daniel Taylor dtay...@vocalabs.comwrote:

  I don't know what platform you are on, but if you are on Linux (and
 possibly BSD) you could use fail2ban to block them at the network
 interface.

   I second fail2ban. If you need some ideas to configure it, you can
 steal them from the freepbx setup.

   How many sip phones do you have outside your network? If few and in
 well-known IPs, consider limiting access to only those (and the sip
 provider you are using).



 On 04/04/2014 09:00 AM, motty cruz wrote:

 Hello All, my asterisk server is constantly under attack

 [Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673
 handle_request_register: Registration from '4941 sip:4941@public_ip'
 failed for '194.100.46.132194.100.46.132:56714' - Wrong password
 [Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673
 handle_request_register: Registration from '4941 sip:4941@public_ip'
 failed for '194.100.46.132194.100.46.132:56714' - Wrong password
 [Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673
 handle_request_register: Registration from '4941 sip:4941@public_ip'
 failed for '194.100.46.132194.100.46.132:56714' - Wrong password
 [Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673
 handle_request_register: Registration from '4941 sip:4941@public_ip'
 failed for '194.100.46.132194.100.46.132:56714' - Wrong password
 [Apr  4 06:56:00] NOTICE[21745]: chan_sip.c:25673
 handle_request_register: Registration from '4941 sip:4941@public_ip'
 failed for '194.100.46.132194.100.46.132:56714' - Wrong password

  is there a way to reject their registration after a three consecutive
 tries?

  Thanks,
  Call
 Send SMS
 Add to Skype
 You'll need Skype CreditFree via Skype




 --
 Daniel Taylor  VP OperationsVocal Laboratories, 
 inc.dtay...@vocalabs.com   http://www.vocalabs.com/(612)235-5711


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Re: [asterisk-users] Asterisk 1.6

[motty cruz]

absolutely right A J, thanks for the heads up.
I do not intent to implement that solution in production server, I hope to
learn it first, build a test server and monitor for a few days or weeks.

Thanks again,


On Fri, Apr 4, 2014 at 7:38 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:

 On Friday 04 Apr 2014, Michelle Dupuis wrote:
  Take a look a SecAst from www.generationd.com
 http://www.generationd.com/
 
  It does everything fail2ban does and more, including blocking users by
  geography (we exclude all of Asia and Africa), detection of break-in
  patterns (even if someone guessed your un/pw), detect changes in dial
  rates, etc.
 
  Grab the free version - its a BIG step up from fail2ban.

 That link points towards a precompiled binary, which could have literally
 *anything* lurking in it.  I politely advise you to back away slowly, and
 break into a run when you think you are out of sight.

 Precompiled binaries without Source Code should be treated like a bottle of
 glowing green liquid labelled drink me, or an offer to come and look at
 some
 puppies.  No reputable software supplier would object to showing you what
 is
 on the inside.

 --
 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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Re: [asterisk-users] Asterisk 1.6

[motty cruz]

Hello Ishfaq, outside users usually travel around the country and connect
from different network, so it won't be possible to lock it down to specific
IP.

Thanks for your support.


On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:




 On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote:

 thank you all for your support. I am using Linux, I only have about 7
 users outside our home network. I will learn fail2ban and will use it
 accordingly.

 again Thanks for your support.



 Do the 7 users outside of your home network always connect from the same
 IP addresses? If so, you can just lock down your SIP port to those 7 IPs
 explicitly in your IPTables configuration.

 Another option would be to change which port you're running SIP on.


 --

 Ishfaq Malik
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
 37 Ducie Street
 Manchester, M1 2JW
 COMPANY REG NO. 04920552


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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk 1.6

[motty cruz]

that sounds feasible, Thanks Michelle,




On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote:

  If you know your users are all from with your country, or state, or even
 city, you could restrict geographic access in your secast.conf file like
 this:


  ruledefault=deny
  ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA

  The above would:
 - By default deny all source IP's anywhere in the world
 - Let in only source IP's from:
 1. North America (continent), Canada (country), Ontario (region)
 2. North America (continent), USA (country), Michigan (region), Detroit
 (city)
 3. Any region called 'Ohio' anywhere in the world (not sure why you would
 do that but fun example)
 4. Anywhere in North America

  So you can open up your system based solely on where you know your real
 users are located.

  -=Michelle=-


  --
 *From:* asterisk-users-boun...@lists.digium.com 
 asterisk-users-boun...@lists.digium.com on behalf of motty cruz 
 motty.c...@gmail.com
 *Sent:* Friday, April 4, 2014 11:15 AM

 *To:* Asterisk Users List
 *Subject:* Re: [asterisk-users] Asterisk 1.6

  Hello Ishfaq, outside users usually travel around the country and
 connect from different network, so it won't be possible to lock it down to
 specific IP.

  Thanks for your support.


 On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:




  On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote:

 thank you all for your support. I am using Linux, I only have about 7
 users outside our home network. I will learn fail2ban and will use it
 accordingly.

  again Thanks for your support.



Do the 7 users outside of your home network always connect from the
 same IP addresses? If so, you can just lock down your SIP port to those 7
 IPs explicitly in your IPTables configuration.

  Another option would be to change which port you're running SIP on.


  --

 Ishfaq Malik
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
 37 Ducie Street
 Manchester, M1 2JW
 COMPANY REG NO. 04920552


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Re: [asterisk-users] callerid overwrite

[motty cruz]

look like the issue continues, I am unable to overwrite callerid from
sip.conf in extensions.conf,

In sip.conf under
[general]
trustrpid = no  should i change it to yes?

Thanks




On Tue, Jan 28, 2014 at 1:06 PM, motty cruz motty.c...@gmail.com wrote:

 Thank you for your reply, I updated extensions.conf file to reflect your
 suggestion, I will monitor Asterisk for any more issues,

 Thanks,



 On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:

  On 1/28/14, 1:55 PM, motty cruz wrote:

  Hi all,
 I'm having issues with overwrite caller id, when I call someone my caller
 id should be mycompanyinc but instead my id shows up as my extension
 number 101.

  this is what i have in sip.conf
  [101]
 type=friend
 context=sipphones
 call-limit=99
 callerid=iuser 101
 disallow=all
 allow=ulaw
 allow=alaw
 username=101
 secret=Passwd
 dtmfmode=rfc2833
 host=dynamic
 mailbox=101@default
 nat=yes
 canreinvite=no


  this is what i have in extensions.conf
 [outbound]
 exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)

 This is how we have it and it works fine on Asterisk 1.8:
 Set(CALLERID(number)=insert your number here)

  exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
 exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
 exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

  any ideas? as this happens random,





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[asterisk-users] callerid overwrite

[motty cruz]

Hi all,
I'm having issues with overwrite caller id, when I call someone my caller
id should be mycompanyinc but instead my id shows up as my extension
number 101.

this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no


this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

any ideas? as this happens random,
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Re: [asterisk-users] callerid overwrite

[motty cruz]

Thank you for your reply, I updated extensions.conf file to reflect your
suggestion, I will monitor Asterisk for any more issues,

Thanks,



On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:

  On 1/28/14, 1:55 PM, motty cruz wrote:

  Hi all,
 I'm having issues with overwrite caller id, when I call someone my caller
 id should be mycompanyinc but instead my id shows up as my extension
 number 101.

  this is what i have in sip.conf
  [101]
 type=friend
 context=sipphones
 call-limit=99
 callerid=iuser 101
 disallow=all
 allow=ulaw
 allow=alaw
 username=101
 secret=Passwd
 dtmfmode=rfc2833
 host=dynamic
 mailbox=101@default
 nat=yes
 canreinvite=no


  this is what i have in extensions.conf
 [outbound]
 exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)

 This is how we have it and it works fine on Asterisk 1.8:
 Set(CALLERID(number)=insert your number here)

  exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
 exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
 exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

  any ideas? as this happens random,





 --
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[asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450

[motty cruz]

Hello, I'm having issues with my phone Polycom sp450 not subscribing to
Asterisk server. Asterisk server is fine, firewall is not the issue because
a secondary phone is working fine, my connection to the server is fine too,
any ideas or suggestions are welcome.

-Motty
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Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450

[motty cruz]

Thank you all for your prompt reply,

my phone was working up until this morning it just stop subscribing to the
Asterisk server.

Version: 3.2.4.0244
phone is configure to download configuration via ftp, again it configure
right because it was working fine.

the phone icon next to the extension number is dark same as the background
so that means is not subscribing to the Asterisk server.

Thank you very much.


On Thu, Jan 2, 2014 at 8:19 AM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:

 asterisk-users-boun...@lists.digium.com wrote on 01/02/2014 10:03:19 AM:

  From: motty cruz motty.c...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com,
  Date: 01/02/2014 10:02 AM
  Subject: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
  Sent by: asterisk-users-boun...@lists.digium.com
 
  Hello, I'm having issues with my phone Polycom sp450 not subscribing
  to Asterisk server. Asterisk server is fine, firewall is not the
  issue because a secondary phone is working fine, my connection to
  the server is fine too, any ideas or suggestions are welcome.
 
  -Motty

 We use Polycom 450s as the main desk phone throughout our company and have
 no issues with them registering. Without seeing your configs, it is hard to
 give you specific advice. The part of your configs you need to be looking
 at (assuming you are provisioning from http or ftp), are the following:

 For the Polycom look at the reg section:
   reg reg.1.displayName=Bob Smith reg.1.address=6534
 reg.1.auth.userId=6534 reg.1.auth.password=myPassword/reg

 For Asterisk (in your sip.conf or other appropriate config file):

   [6534](polycom)
   callerid=Bob SMith 6534
   secret=myPassword
   mailbox=6534

 A couple of notes here: 6534 is the extension number for Bob Smith.
 myPassword in the files should be replaced with whatever password you
 have assigned for that phone. The (polycom) template contains all the
 options needed for phones to work in my specific install, but doesn't have
 anything that would affect registration.

 If you watch the asterisk console when you boot up the phone, do you get
 any errors in the console? I know when I am testing/experimenting with new
 setups that I often see errors when the phone goes to register. It usually
 is because I have either specified a username that doesn't exist in
 Asterisk or I have the phone passing an incorrect password with what is
 specified in sip.conf.
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Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450

[motty cruz]

Thank you all,

After setting the phone to factory defaults, entered configuration
parameters, phone is working again. I really don't know why all sudden stop
working. at least know i have a working phone I will go thoroughly through
the logs, I hope to find the answer, if I do I will post it here.

Thank you again.


On Thu, Jan 2, 2014 at 9:12 AM, Ryan Wagoner rswago...@gmail.com wrote:


 On Thu, Jan 2, 2014 at 11:13 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Which firmware version?  4.1.x is only for use with MS Link server.  A
 symptom of running 4.1.x firmware with a non-MS server is the phone will
 not show buddies.


 I'm running 4.1.0 on a Polycom IP 335 and IP 550 and version 4.1.5 on a
 Polycom VVX 400. Buddies work on all three phones. The firmware is for both
 SIP and Lync. You change the base profile option accordingly. Look in the
 Polycom UC Software Admin Guide for more information.

 Ryan


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[asterisk-users] Asterisk 1.8.22

[motty cruz]

Hello, I have a fully functional Asterisk Server, I want to configure this
server to be able to process call from Skype, can someone point me to a
howto? or if there are suggestions on best way to approach this problem.

Thanks,
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Re: [asterisk-users] Asterisk 1.8.22

[motty cruz]

Hello Mitul, I did purchase a sip channel from skype, and configure it on
Asterisk

in Sip.conf
register = motty2342342:mypass...@sip.skype.com/motty2342342

[skype]
username=motty2342342
secret=mypass123
type=friend
qualify=yes
context=from-skype
host=sip.skype.com

in a testing skype account client I see is off line, in the manager
daskboard in skype sip profiles was register successfuly.


I don't know what else to do, any ideas?


On Fri, Nov 8, 2013 at 9:30 AM, Mitul Limbani mi...@enterux.in wrote:

 Buy SIP Channel from Skype and you can configure it as sip trunk on
 asterisk box.

 Mitul


 On Friday, November 8, 2013, motty cruz wrote:

 Hello, I have a fully functional Asterisk Server, I want to configure
 this server to be able to process call from Skype, can someone point me to
 a howto? or if there are suggestions on best way to approach this problem.

 Thanks,



 --
 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-71967196
 Cell: +91-9820332422



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[asterisk-users] asterisk 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic

[motty cruz]

Hello All,

I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get
meetme feature to work when dial meetme extension, can you please help?

It always worked before, also I do not have dahdi installed on this
machine, never did.

   -- Executing [104@sipphones:1] MeetMe(SIP/101-0813, 104) in new
stack
  == Parsing '/etc/asterisk/meetme.conf':   == Found
[Jun  6 13:17:30] WARNING[11457]: app_meetme.c:1248 build_conf: Unable to
open DAHDI pseudo device


Thanks,
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Re: [asterisk-users] asterisk 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic

[motty cruz]

Thanks Johan,

I did noticed /etc/dahdi so you're right it was installed on one point, I
re-install dahdi and problem went away.

Thank you very much!


On Thu, Jun 6, 2013 at 1:45 PM, Johan Wilfer li...@jttech.se wrote:

 2013-06-06 22:21, motty cruz skrev:
  Hello All,
 
  I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't
  get meetme feature to work when dial meetme extension, can you please
 help?
 
  It always worked before, also I do not have dahdi installed on this
  machine, never did.
 
 -- Executing [104@sipphones:1] MeetMe(SIP/101-0813, 104) in
  new stack
== Parsing '/etc/asterisk/meetme.conf':   == Found
  [Jun  6 13:17:30] WARNING[11457]: app_meetme.c:1248 build_conf: Unable
  to open DAHDI pseudo device
 

 Meetme is depending on DAHDI, and have always been. Asterisk would not
 even compile Meetme if DAHDI header files isn't present.

 So you must have installed DAHDI at some point?

 Try run dahdi_test (after installing), you should get output like 99.9%
 99.8% ... and so on. Then you know the timing works (dahdi psuedo)


 --
 Johan Wilfer


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[asterisk-users] monitoring Asterisk 1.8

[motty cruz]

Hello,

i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
but no success, I do prefer not to install any web server on the server
running Asterisk.


Thanks in advance.
-Motty
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Re: [asterisk-users] monitoring Asterisk 1.8

[motty cruz]

Thanks for the suggestion Carlos,

do you have a HowTo? can you point me to one.

I unsuccessfully follow one found using google. I'm using CentOs 6.0

Thanks,
Motty


On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote:

 Monitor what parts exactly?

 Right this moment I'm in the process of installing Munin and the Asterisk
 plugin to monitor channel usage, SIP connections, and the like.  The Munin
 server is running on a separate machine with just the node software on
 Asterisk.



 On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed
 Nagios but no success, I do prefer not to install any web server on the
 server running Asterisk.


 Thanks in advance.
 -Motty

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 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


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Re: [asterisk-users] monitoring Asterisk 1.8

[motty cruz]

Thanks for your help; I just want to monitor the queue, calls on hold
average time, incoming out going call, I only want to monitor Asterisk, not
the server Asterisk in running on.

thanks,
-Motty


On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 http://opennms.org/wiki/Installation:Yum


 On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 I'm using opennms and It's working fine.





 On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed
 Nagios but no success, I do prefer not to install any web server on the
 server running Asterisk.


 Thanks in advance.
 -Motty

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Re: [asterisk-users] asterisk 1.8.10.1 meetme

[motty cruz]

Hello Jonathan,
I thank you for prompt reply to my post.

I'm using SIP trunks with Polycom sp450 devices.

Also, I was wrong to mention meetme, my conference does not involve using
meetme feature on Asterisk.

It does not happen often, it happens random.


On Thu, Feb 7, 2013 at 3:16 PM, Jonathan Rose jr...@digium.com wrote:

 motty cruz wrote:
  Hello,
  I'm running Asterisk 1.8.10 on Linux box, when I'm in a
  conference(meetme) with another person, and a third person join our
  conference when the third person leave the conference I get
  disconnected from the original conference with a second party. I
  hope this clear.
  This does not happen often, is random, anybody experience something
  similar? or any idea how to fix this problem?

 Let me just start by saying that MeetMe has been touched by a rather
 large number of patches in the 11 months and it's quite likely that
 your problem will be fixed if you upgrade. r373242 comes to mind in
 particular.

 Other than that though, it would be helpful if you added some
 additional information, such as what arguments are are running meetme
 with and what kinds of devices you are connecting with (SIP phones
 presumably?)



 --
 Jonathan R. Rose
 Digium, Inc. | Software Engineer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 direct +1 256 428 6139

 Check us out at: http://digium.com  http://asterisk.org

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[asterisk-users] asterisk 1.8.10.1 meetme

[motty cruz]

Hello,
I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme)
with another person, and a third person join our conference when the third
person leave the conference I get disconnected from the original conference
with a second party. I hope this clear.
This does not happen often, is random, anybody experience something
similar? or any idea how to fix this problem?

Thanks,
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Re: [asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily

[motty cruz]

I have Polycom IP550. The Forward No Answer is working fine when
enabled. I was looking at the sip.cfg but don't know exactly what to look
for, can you give me a hint to where would i find that option?

Thanks,

On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill 
justin.sherr...@americanrocksalt.com wrote:

 I have several Polycom IP550 phones running UC 4.0.3, connected to
 Asterisk 1.8.

 Setting forwarding for Always  works as expected; the phone issues a 302
 Moved Temporarily, and Asterisk shifts the call to the new location.

 Setting forwarding to No Answer means a 302 never gets issued.  It just
 rings and eventually goes to voicemail.  Watching with Wireshark, I never
 see a 302 SIP message issued.  I can't find anything in the phone settings
 that look like it would disable this.

 Anyone else with a Polycom set that sees this, or does not see this and
 has forward no answer working?

 Justin Sherrill - American Rock Salt
 P: 585-991-6825 F: 585-991-6925 C: 585-298-6826


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