date:20161013

Re: [asterisk-users] Asterisk inside network. What phone works well?

2016-10-13 Thread kevin . larsen

> I have Asterisk running well inside our network. I did some 
> experiments exposing it to internet but had some issues:
> 1. NAT issues (voice one way, etc). From what I understand double-
> NAT users will always have something like this
> 2. Immediately I see people trying to hack into. I did configure 
> Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc
> 
> So.. I ended up closing network. Currently most users inside 
> network. My home router have GRE tunnel to office so phone works just 
fine.
> Another user uses VPN and soft phone and it works good too.
> 
> Now I need to setup some users with actual phone devices and none of
> those solutions will work. So, I did some research and found 
> that some phones have VPN capability built in. 
> 
> Right now I use Cisco SPA504G phones. We have auto-provisioning for 
> them, works well. But I don’t think they have VPN capability.
> 
> 
> What I found it that Cisco 525g2 has AnyConnect functionality (SSL 
> VPN) but not sure if this is what I need.
> 
> We have Mikrotik router. Can I setup VPN on router and have this 
> Cisco phone auto-dial VPN and then connect to Asterisk? I’m asking 
> to see if this will work before I go in and buy that phone.
> Or maybe there is other devices/solutions you suggest? I’d like to 
> stay with Cisco because I’m somewhat familiar with provisioning those..

I haven't done this myself, but I think what you need to look at is phones 
that can do IPSEC vpn setups.

For the Mikrotik router, this may be helpful to start investigating:
http://wiki.mikrotik.com/wiki/L2TP_%2B_IPSEC_between_Mikrotik_router_and_a_PC

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[asterisk-users] Asterisk inside network. What phone works well?

2016-10-13 Thread Ivan Demkovitch

Hello list,

I have Asterisk running well inside our network. I did some experiments 
exposing it to internet but had some issues:
1. NAT issues (voice one way, etc). From what I understand double-NAT users 
will always have something like this
2. Immediately I see people trying to hack into. I did configure Fail2Ban and 
it works somewhat, but not 100%. Erroneous logs, etc

So.. I ended up closing network. Currently most users inside network. My home 
router have GRE tunnel to office so phone works just fine.
Another user uses VPN and soft phone and it works good too.

Now I need to setup some users with actual phone devices and none of those 
solutions will work. So, I did some research and found 
that some phones have VPN capability built in. 

Right now I use Cisco SPA504G phones. We have auto-provisioning for them, works 
well. But I don’t think they have VPN capability.


What I found it that Cisco 525g2 has AnyConnect functionality (SSL VPN) but not 
sure if this is what I need.

We have Mikrotik router. Can I setup VPN on router and have this Cisco phone 
auto-dial VPN and then connect to Asterisk? I’m asking to see if this will work 
before I go in and buy that phone.
Or maybe there is other devices/solutions you suggest? I’d like to stay with 
Cisco because I’m somewhat familiar with provisioning those..

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

2016-10-13 Thread Victor Villarreal

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"  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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Re: [asterisk-users] Openfile Issue

2016-10-13 Thread Ahmed Munir

[root@abc asterisk]# lsof -u 50771 | wc -l
0

BTW, I'm using CentOS 6.5



>
> Date: Thu, 13 Oct 2016 10:20:19 -0400
>> From: Dovid Bender 
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Subject: Re: [asterisk-users] Openfile Issue
>> Message-ID:
>>

[asterisk-users] Wildcard AEX800 digium card asterisk configuration

2016-10-13 Thread christopher kamutumwa

hello,

i recently purchased a Wildcard AEX800 digium card. Ive installed
asterisk 13 and all prerequistses on ubuntu serv14.04 LTS. Dahi is the
driver am using; ive configured all but when i call from PSTN through
fxo port an not getting anything in logs or to extensions. below are
my config

please help

system.conf

loadzone = za
defaultzone =za
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4
fxoks=5
echocanceller=mg2,5
fxoks=6
echocanceller=mg2,6
fxoks=7
echocanceller=mg2,7
fxoks=8
echocanceller=mg2,8



chan dahdi.conf

[trunkgroups]

; No trunk groups are needed in this configuration.

[channels]
#include /etc/asterisk/dahdi-channels.conf
; The channels context is used when defining channels

using the
; older deprecated method.  Don't use this as a section

name.

;[phone](!)
;
; A template to hold common options for all phones.
;
usecallerid = yes
hidecallerid = no
callwaiting = yes
threewaycalling = yes
transfer = yes
echocancel = yes
echocancelwhenbridged = yes
;immediate = no
rxgain = 0.0
txgain = 0.0
;FXS Modules
group = 1
echocancel = yes
signalling = fxo_ks
context = Internal
channel = 1-4

;FXO Modules
group = 2
echocancel = yes
signalling = fxs_ks
context = Incoming
channel = 5-8


voicemail.conf

[default]

1234 => 4242,chris kamutumwa,root@localhost

1000 => 1234,chris kamutumwa,chriskamutu...@gmail.com
2000 => 1234,chris utumwa,ch...@crystaline.co.zm

~
~

extension.conf

[Internal]
exten => 1000,1,Dial (DAHDI/1,20,rt)
exten => 1000,2,Voicemail (1000,u)
exten => 1000,102,Voicemail (1000,b)


exten => 2000,1,Dial (DAHDI/2,20,rt)
exten => 2000,2,Voicemail (2000,u)
exten => 2000,102,Voicemail (2000,b)


exten => 8500,1,VoiceMailMain
exten => 8501,1,MusicOnHold

exten => _9.,1,Dail (DAHDI/g2/www${EXTEN:1} )
exten => _9.,2,Congestion

[Incoming]
exten => s,1,Answer
exten => s,2,Dial(DAHDI/g1,20,rt)
exten => s,3,Voicemail(1000,u)
exten => s,103,Voicemail(1000,b)


root@ubuntu:/etc/asterisk# lsdahdi
### Span  1: WCTDM/0 "Wildcard AEX800" (MASTER)
  1 FXO RED
  2 FXO RED
  3 FXO RED
  4 FXO RED
  5 FXS
  6 FXS
  7 FXS
  8 FXS
root@ubuntu:/etc/asterisk# dahdi_cfg -vvv
DAHDI Tools Version - 2.11.1

DAHDI Version: 2.11.1
Echo Canceller(s):
Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
Channel 05: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05)
Channel 06: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06)
Channel 07: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07)
Channel 08: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08)

8 channels to configure.

Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
Setting echocan for channel 3 to mg2
Setting echocan for channel 4 to mg2
Setting echocan for channel 5 to mg2
Setting echocan for channel 6 to mg2
Setting echocan for channel 7 to mg2
Setting echocan for channel 8 to mg2

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Re: [asterisk-users] Openfile Issue

2016-10-13 Thread Dovid Bender

50771 is the PID. I am talking about the user. for instances if running as
root (which you should never do) then:
 lsof -u root | wc -l

On Thu, Oct 13, 2016 at 1:31 PM, Ahmed Munir 
wrote:

>
> [root@abc asterisk]# lsof -u 50771 | wc -l
> 0
>
> BTW, I'm using CentOS 6.5
>
>
>
>>
>> Date: Thu, 13 Oct 2016 10:20:19 -0400
>>> From: Dovid Bender 
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> 
>>> Subject: Re: [asterisk-users] Openfile Issue
>>> Message-ID:
>>>

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

2016-10-13 Thread 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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Re: [asterisk-users] Asterisk inside network. What phone works well?

2016-10-13 Thread Mark Wiater

I think you had asked what phone works well with VPN's. I've had very 
good experiences with Yealink using OpenVPN, never an issue.


I think I've heard that Snom does OpenVPN as well.

Mark

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

2016-10-13 Thread 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" ;tag=2859342B-CBC71460

SIP Display info: "1006"

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

SIP from tag: 2859342B-CBC71460

To: 

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: ;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"  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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Re: [asterisk-users] Openfile Issue

2016-10-13 Thread Steve Edwards


On Thu, 13 Oct 2016, Dovid Bender wrote:


50771 is the PID. I am talking about the user. for instances if running
as root (which you should never do) then:



 lsof -u root | wc -l


Wouldn't

sudo lsof -c asterisk | wc --lines

be more direct and accurate?

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[asterisk-users] Asterisk 14.0.2 opens a high numbered UDP port

2016-10-13 Thread Brandon B.

What part of Asterisk 14.0.2 opens the random, high numbered (33094 
currently) UDP port? This port is opened even without any channel 
drivers loaded.


$ sudo netstat -ltunp | grep asterisk
udp0  0 0.0.0.0:51488 
0.0.0.0:*   13830/asterisk
udp0  0 0.0.0.0:5060 0.0.0.0:*   
13830/asterisk
udp0  0 :::42516 :::*
13830/asterisk




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

2016-10-13 Thread Victor Villarreal

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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Re: [asterisk-users] Asterisk 14.0.2 opens a high numbered UDP port

2016-10-13 Thread George Joseph

On Thu, Oct 13, 2016 at 12:35 PM, Brandon B.  wrote:

> What part of Asterisk 14.0.2 opens the random, high numbered (33094
> currently) UDP port? This port is opened even without any channel drivers
> loaded.
>
> $ sudo netstat -ltunp | grep asterisk
> udp0  0 0.0.0.0:51488 0.0.0.0:*
>  13830/asterisk
> udp0  0 0.0.0.0:5060 0.0.0.0:*
>  13830/asterisk
> udp0  0 :::42516 :::*
> 13830/asterisk
>
>
>
Those ports are used by the underlying pjproject DNS resolver.  The
resolver is always listening on those ports for DNS query responses.  1 for
IPV4 and 1 for IPV6.  Your firewall should only be allowing responses to
flow through to those ports that match outgoing requests.





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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Asterisk 12 error when installing

2016-10-13 Thread Jonathan H

Are those numbers correct?

Asterisk 12 stopped being supported almost 2 years ago and became "do
not use" on 2015-12-20
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Ubuntu 14 may still be supported, if you're on 14.0.4.5
https://wiki.ubuntu.com/Releases

You could try make distclean and start again.

What happens when you compile with a current, supported version of Asterisk?


On 13 October 2016 at 08:33, christopher kamutumwa
 wrote:
> Hello,
>
> Am trying to install asterisk 12 on ubuntu 14.04lts and am getting the
> below error after a MAKE any hints how to go round this?
>
> bedit.a  -> asterisk
> asterisk.o: file not recognized: File truncated
> collect2: error: ld returned 1 exit status
> make[1]: *** [asterisk] Error 1
> make: *** [main] Error 2
>
>
> regards
>
> chris
>
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[asterisk-users] Asterisk 12 error when installing

2016-10-13 Thread christopher kamutumwa

Hello,

Am trying to install asterisk 12 on ubuntu 14.04lts and am getting the
below error after a MAKE any hints how to go round this?

bedit.a  -> asterisk
asterisk.o: file not recognized: File truncated
collect2: error: ld returned 1 exit status
make[1]: *** [asterisk] Error 1
make: *** [main] Error 2


regards

chris

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[asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

2016-10-13 Thread Jonathan H

Back to basics here. I want to match on one OR two digits.

The following two both work, but ONLY for more than one digit, which
is not as expected from the docs (see below).

exten => _X.,1,SayNumber(${EXTEN})
exten => _[0-9].,1,SayNumber(${EXTEN})


This next one will ONLY match 2 digits, as expected, but the first two
SHOULD match one or more, right?

exten => _XX,1,SayNumber(${EXTEN})

The following pattern works, but I thought it was "dangerous" and to
be discouraged?
exten => _.,1,SayNumber(${EXTEN})

So, again, if someone dials 1 and a one second delay passes, I want it to say 1.
If someone dials 1 then another 1 within a second then I want it to be
11, and 111 should be invalid.

(I've Set(TIMEOUT(digit)=1) )

Yes, I can do this with multiple lines, but the docs suggest this
should be easily do-able in 1 line, and I don't want to double the
amount of dialplan (there'll be a few of these!).

Here are my references:

---

https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching

The letter X or x represents a single digit from 0 to 9.
The period character (.) at the end of a pattern matches one or more
remaining characters. You put it at the end of a pattern when you want
to match extensions of an indeterminate length.

---

Page 141 of the Asterisk Definitive Guide 4th Edition:

. (period)
Wildcard match; matches one or more characters, no matter what they are.
If you’re not careful, wildcard matches can make your dialplans do
things you’re not expecting (like matching built-in extensions such
as i or h). You should use the wildcard match in a pattern only after
you’ve matched as many other digits as possible. For example, the
following pattern match should probably never be used:
_.
In fact, Asterisk will warn you if you try to use it. Instead, if you
really need a catchall pattern match, use this one to match all strings
that start with a digit followed by one or more characters (see ! if
you want to be able to match on zero or more characters):
_X.
Or this one, to match any alphanumeric string:
_[0-9a-zA-Z].

---

http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
Do not use a pattern of _. as this will match everything including
Asterisk special extensions like i, t, h, etc. Instead use something
like _X. or _X which will not match __special__ extensions..
So what do you use instead of _. ? Many examples use this construct,
but if you use it you may see a warning message in the log advising
you to change _. to _X.

---

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Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

2016-10-13 Thread Jean Aunis


You can use the "!" character :

exten => _X!,1,SayNumber(${EXTEN})


Best regards

Jean Aunis

Le 13/10/2016 à 12:54, Jonathan H a écrit :

Back to basics here. I want to match on one OR two digits.

The following two both work, but ONLY for more than one digit, which
is not as expected from the docs (see below).

exten => _X.,1,SayNumber(${EXTEN})
exten => _[0-9].,1,SayNumber(${EXTEN})


This next one will ONLY match 2 digits, as expected, but the first two
SHOULD match one or more, right?

exten => _XX,1,SayNumber(${EXTEN})

The following pattern works, but I thought it was "dangerous" and to
be discouraged?
exten => _.,1,SayNumber(${EXTEN})

So, again, if someone dials 1 and a one second delay passes, I want it to say 1.
If someone dials 1 then another 1 within a second then I want it to be
11, and 111 should be invalid.

(I've Set(TIMEOUT(digit)=1) )

Yes, I can do this with multiple lines, but the docs suggest this
should be easily do-able in 1 line, and I don't want to double the
amount of dialplan (there'll be a few of these!).

Here are my references:

---

https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching

The letter X or x represents a single digit from 0 to 9.
The period character (.) at the end of a pattern matches one or more
remaining characters. You put it at the end of a pattern when you want
to match extensions of an indeterminate length.

---

Page 141 of the Asterisk Definitive Guide 4th Edition:

. (period)
Wildcard match; matches one or more characters, no matter what they are.
If you’re not careful, wildcard matches can make your dialplans do
things you’re not expecting (like matching built-in extensions such
as i or h). You should use the wildcard match in a pattern only after
you’ve matched as many other digits as possible. For example, the
following pattern match should probably never be used:
_.
In fact, Asterisk will warn you if you try to use it. Instead, if you
really need a catchall pattern match, use this one to match all strings
that start with a digit followed by one or more characters (see ! if
you want to be able to match on zero or more characters):
_X.
Or this one, to match any alphanumeric string:
_[0-9a-zA-Z].

---

http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
Do not use a pattern of _. as this will match everything including
Asterisk special extensions like i, t, h, etc. Instead use something
like _X. or _X which will not match __special__ extensions..
So what do you use instead of _. ? Many examples use this construct,
but if you use it you may see a warning message in the log advising
you to change _. to _X.

---




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Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

2016-10-13 Thread Jonathan H

Sorry, I should have said, I already tried "!".

It matches immediately and doesn't wait for a second digit.

On 13 October 2016 at 12:05, Jean Aunis  wrote:
> You can use the "!" character :
>
> exten => _X!,1,SayNumber(${EXTEN})
>
>
> Best regards
>
> Jean Aunis
>
>
> Le 13/10/2016 à 12:54, Jonathan H a écrit :
>>
>> Back to basics here. I want to match on one OR two digits.
>>
>> The following two both work, but ONLY for more than one digit, which
>> is not as expected from the docs (see below).
>>
>> exten => _X.,1,SayNumber(${EXTEN})
>> exten => _[0-9].,1,SayNumber(${EXTEN})
>>
>>
>> This next one will ONLY match 2 digits, as expected, but the first two
>> SHOULD match one or more, right?
>>
>> exten => _XX,1,SayNumber(${EXTEN})
>>
>> The following pattern works, but I thought it was "dangerous" and to
>> be discouraged?
>> exten => _.,1,SayNumber(${EXTEN})
>>
>> So, again, if someone dials 1 and a one second delay passes, I want it to
>> say 1.
>> If someone dials 1 then another 1 within a second then I want it to be
>> 11, and 111 should be invalid.
>>
>> (I've Set(TIMEOUT(digit)=1) )
>>
>> Yes, I can do this with multiple lines, but the docs suggest this
>> should be easily do-able in 1 line, and I don't want to double the
>> amount of dialplan (there'll be a few of these!).
>>
>> Here are my references:
>>
>> ---
>>
>> https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
>>
>> The letter X or x represents a single digit from 0 to 9.
>> The period character (.) at the end of a pattern matches one or more
>> remaining characters. You put it at the end of a pattern when you want
>> to match extensions of an indeterminate length.
>>
>> ---
>>
>> Page 141 of the Asterisk Definitive Guide 4th Edition:
>>
>> . (period)
>> Wildcard match; matches one or more characters, no matter what they are.
>> If you’re not careful, wildcard matches can make your dialplans do
>> things you’re not expecting (like matching built-in extensions such
>> as i or h). You should use the wildcard match in a pattern only after
>> you’ve matched as many other digits as possible. For example, the
>> following pattern match should probably never be used:
>> _.
>> In fact, Asterisk will warn you if you try to use it. Instead, if you
>> really need a catchall pattern match, use this one to match all strings
>> that start with a digit followed by one or more characters (see ! if
>> you want to be able to match on zero or more characters):
>> _X.
>> Or this one, to match any alphanumeric string:
>> _[0-9a-zA-Z].
>>
>> ---
>>
>> http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
>> Do not use a pattern of _. as this will match everything including
>> Asterisk special extensions like i, t, h, etc. Instead use something
>> like _X. or _X which will not match __special__ extensions..
>> So what do you use instead of _. ? Many examples use this construct,
>> but if you use it you may see a warning message in the log advising
>> you to change _. to _X.
>>
>> ---
>>
>
>
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[asterisk-users] Softphone with by-default AutoAnswer active?

2016-10-13 Thread Mandar Khire

Hi,
I know that This is not Asterisk related question but then also I ask this
question here due to Asterisk users know about softphones & here lots of
user present.

Question:-

I am looking for Softphone which work on Windows platform.

Softphone must have 'default AutoAnswer on'.

Means example:- When I install softphone, it does not have any SIP
registration. But it has some by-default settings.

In that I am looking active AutoAnswer option.

I tried Ekiga, Linphone, mizuphone, x-lite etc but all have default
AutoAnswer off.

I have to click on settings & active it.

So all these softphones not useful for me.

Need help.
Thanks,
Mandar P. Khire
+919769419340
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Re: [asterisk-users] Softphone with by-default AutoAnswer active?

2016-10-13 Thread Dan Jenkins

On Thu, Oct 13, 2016 at 12:19 PM, Mandar Khire 
wrote:

> Hi,
> I know that This is not Asterisk related question but then also I ask this
> question here due to Asterisk users know about softphones & here lots of
> user present.
>
> Question:-
>
> I am looking for Softphone which work on Windows platform.
>
> Softphone must have 'default AutoAnswer on'.
>
> Means example:- When I install softphone, it does not have any SIP
> registration. But it has some by-default settings.
>
> In that I am looking active AutoAnswer option.
>
> I tried Ekiga, Linphone, mizuphone, x-lite etc but all have default
> AutoAnswer off.
>
> I have to click on settings & active it.
>
> So all these softphones not useful for me.
>
> Need help.
> Thanks,
> Mandar P. Khire
> +919769419340
>
>
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>


>From what I remember Bria allows this either by a setting on the OS or via
a configuration server which provisions the settings on login
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Re: [asterisk-users] Openfile Issue

2016-10-13 Thread Dovid Bender

What do you get when you do:
cat /proc//limits ?

On Thu, Oct 13, 2016 at 9:30 AM, Ahmed Munir 
wrote:

> Hi all,
>
> Now a days getting openfile issues on asterisk quite often even setting
> system  soft limit to 2 and hard limit to 25000 and issue usually
> occurs during openfile socket consumed by system and asterisk is quite
> smaller than the soft or hard limit. See below system and asterisk logs;
>
> 2016:10:13_08:19:01 | Too many LOG file moved successfully - messages
>
> 2016:10:13_08:19:01 | Asterisk openfile count: 1252
>
> 2016:10:13_08:19:01 | Total system open files count: 4091
>
> 2016:10:13_08:19:01 | SIP channels: 93 active calls|1563 calls processed
>
> 2016:10:13_08:19:01 | Asterisk SIP peers: 366
>
> 2016:10:13_08:19:01 | Asterisk service uptime: System uptime: 4 days, 4
> hours, 12 minutes, 33 seconds
>
> Last reload: 4 days, 4 hours, 12 minutes, 33 seconds
>
> Privilege escalation protection disabled!
>
> See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
>
> 2016:10:13_08:19:01 | Socket Summary
>
> Total: 648 (kernel 758)
>
> TCP:   20 (estab 8, closed 5, orphaned 0, synrecv 0, timewait 1/0), ports
> 14
>
>
>
> Transport Total IPIPv6
>
> * 758   - -
>
> RAW   0 0 0
>
> UDP   422   419   3
>
> TCP   15141
>
> INET  437   433   4
>
> FRAG  0 0 0
>
> 2016:10:13_08:19:01 | Logged successfully all the required details
>
>
> [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c:
> Failed to create timerfd timer: Too many open files
>
> [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket
>
> [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c:
> Failed to create timerfd timer: Too many open files
>
> [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket
>
> [2016-10-13 08:18:41] ERROR[50689][C-0629] res_timing_timerfd.c:
> Failed to create timerfd timer: Too many open files
>
> [2016-10-13 08:18:41] ERROR[50689][C-0629] acl.c: Cannot create socket
>
> [2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to
> re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many
> open files
>
> [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
>
> [2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to
> re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many
> open files
>
> [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
>
> [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
>
> Further added, I'm using CentOS 6.5 as OS.
>
> Please advise what changes required for permanently fixing this random
> issue.
>
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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[asterisk-users] Openfile Issue

2016-10-13 Thread Ahmed Munir

Hi all,

Now a days getting openfile issues on asterisk quite often even setting
system  soft limit to 2 and hard limit to 25000 and issue usually
occurs during openfile socket consumed by system and asterisk is quite
smaller than the soft or hard limit. See below system and asterisk logs;

2016:10:13_08:19:01 | Too many LOG file moved successfully - messages

2016:10:13_08:19:01 | Asterisk openfile count: 1252

2016:10:13_08:19:01 | Total system open files count: 4091

2016:10:13_08:19:01 | SIP channels: 93 active calls|1563 calls processed

2016:10:13_08:19:01 | Asterisk SIP peers: 366

2016:10:13_08:19:01 | Asterisk service uptime: System uptime: 4 days, 4
hours, 12 minutes, 33 seconds

Last reload: 4 days, 4 hours, 12 minutes, 33 seconds

Privilege escalation protection disabled!

See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.

2016:10:13_08:19:01 | Socket Summary

Total: 648 (kernel 758)

TCP:   20 (estab 8, closed 5, orphaned 0, synrecv 0, timewait 1/0), ports 14



Transport Total IPIPv6

* 758   - -

RAW   0 0 0

UDP   422   419   3

TCP   15141

INET  437   433   4

FRAG  0 0 0

2016:10:13_08:19:01 | Logged successfully all the required details


[2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: Failed
to create timerfd timer: Too many open files

[2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket

[2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: Failed
to create timerfd timer: Too many open files

[2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket

[2016-10-13 08:18:41] ERROR[50689][C-0629] res_timing_timerfd.c: Failed
to create timerfd timer: Too many open files

[2016-10-13 08:18:41] ERROR[50689][C-0629] acl.c: Cannot create socket

[2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files

[2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket

[2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files

[2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket

[2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket

Further added, I'm using CentOS 6.5 as OS.

Please advise what changes required for permanently fixing this random
issue.


-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

2016-10-13 Thread Tony Mountifield

In article ,
Jonathan H  wrote:
> Back to basics here. I want to match on one OR two digits.
> 
> The following two both work, but ONLY for more than one digit, which
> is not as expected from the docs (see below).
> 
> exten => _X.,1,SayNumber(${EXTEN})
> exten => _[0-9].,1,SayNumber(${EXTEN})
> 
> 
> This next one will ONLY match 2 digits, as expected, but the first two
> SHOULD match one or more, right?
> 
> exten => _XX,1,SayNumber(${EXTEN})
> 
> The following pattern works, but I thought it was "dangerous" and to
> be discouraged?
> exten => _.,1,SayNumber(${EXTEN})
> 
> So, again, if someone dials 1 and a one second delay passes, I want it to say 
> 1.
> If someone dials 1 then another 1 within a second then I want it to be
> 11, and 111 should be invalid.
> 
> (I've Set(TIMEOUT(digit)=1) )
> 
> Yes, I can do this with multiple lines, but the docs suggest this
> should be easily do-able in 1 line, and I don't want to double the
> amount of dialplan (there'll be a few of these!).

When matching an extension being dialled, Asterisk is only concerned
about priority 1, so that's the only priority you need to double.
You should be able to use ! safely in priority 2 upwards:

exten => _X,1,NoOp(Matching single digit)
exten => _X.,1,NoOp(Matching multiple digits)
exten => _X!,2,SayNumber(${EXTEN})
exten => _X!,3,Etc..

Disclaimer: I haven't tested this.

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] Openfile Issue

2016-10-13 Thread Ahmed Munir

.c: Cannot create socket
> >
> > Further added, I'm using CentOS 6.5 as OS.
> >
> > Please advise what changes required for permanently fixing this random
> > issue.
> >
> >
> > --
> > Regards,
> >
> > Ahmed Munir Chohan
> >
> >
> > --
> > _____
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
> >   http://www.asterisk.org/community/astricon-user-conference
> >
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> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> -- next part --
> An HTML attachment was scrubbed...
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> s/20161013/dd80ddc4/attachment.html>
>
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> End of asterisk-users Digest, Vol 147, Issue 11
> ***
>



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Ahmed Munir Chohan
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Re: [asterisk-users] Openfile Issue

2016-10-13 Thread Dovid Bender

0689][C-0629] res_timing_timerfd.c:
>> > Failed to create timerfd timer: Too many open files
>> >
>> > [2016-10-13 08:18:41] ERROR[50689][C-0629] acl.c: Cannot create
>> socket
>> >
>> > [2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to
>> > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many
>> > open files
>> >
>> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
>> >
>> > [2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to
>> > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many
>> > open files
>> >
>> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
>> >
>> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
>> >
>> > Further added, I'm using CentOS 6.5 as OS.
>> >
>> > Please advise what changes required for permanently fixing this random
>> > issue.
>> >
>> >
>> > --
>> > Regards,
>> >
>> > Ahmed Munir Chohan
>> >
>> >
>> > --
>> > _
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>> >   http://www.asterisk.org/community/astricon-user-conference
>> >
>> > New to Asterisk? Start here:
>> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
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>>
>> ___
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>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>   http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
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>> End of asterisk-users Digest, Vol 147, Issue 11
>> ***
>>
>
>
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk inside network. What phone works well?

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Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

Re: [asterisk-users] Openfile Issue

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