Re: [asterisk-users] Dial() from the console?

2017-01-11 Thread Tzafrir Cohen 
On Wed, Jan 11, 2017 at 07:31:31AM -0500, Doug Lytle wrote:
> >>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote:
> 
> >>> Can I dial directly from the asterisk console with the Dial() application?
> 
> 
> console dial number@context

Note, however, that it is a different thing. It uses a specific device:
the "console". That is: the speaker and microphone of the system
Asterisk is running on. For that to be available you need to have one of
chan_alsa, chan_console or chan_oss loaded and working.

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[asterisk-users] 256 bit SRTP ciphers in Asterisk 14.x , only works for outbound call ?

2017-01-11 Thread Kevin Long 


Greetings,


If I understand correctly,  Asterisk 14 introduced support for some new SRTP 
ciphers (including some 256 bit ones), previously only two 128 bit ciphers were 
supported.

Using Asterisk 14, I was able to make a call from a softphone (Groundwire) with 
a 256 bit cipher suite on SRTP, which is great. 


However,  I don’t see any way to specify with PJSIP (or chan sip) ,  the cipher 
suite which should be used when Asterisk calls the endpoint/phone.


So I believe this means Asterisk would always use 128 bit SRTP to call the 
phone.  Then, if you have 256 bit only ciphers allowed on the phone, the call 
fails.


Perhaps this is just not documented, or may not be implemented yet.  Anyone 
have a thought? 

Thank you,.

Kevin Long
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Re: [asterisk-users] PJSIP status check at DB level

2017-01-11 Thread Joshua Colp 
On Wed, Jan 11, 2017, at 12:59 PM, Ahmed Munir wrote:
> Thanks.
> 
> But I was not able to find the records in 'ps_contacts' table. As per my
> 'ps_aors' entries, I'm enabling and using only following fields below;
> 
> id: 20010 
> max_contacts: 1
> remove_existing: yes
> qualify_frequecy: 10

What is your sorcery.conf? Have you configured it such that your
contacts are put into the database?

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Re: [asterisk-users] PJSIP status check at DB level

2017-01-11 Thread Ahmed Munir 
Thanks.

But I was not able to find the records in 'ps_contacts' table. As per my
'ps_aors' entries, I'm enabling and using only following fields below;

id: 20010 
max_contacts: 1
remove_existing: yes
qualify_frequecy: 10

Please advise if there any more parameters I may need to set in 'ps_aors'
table.


Date: Tue, 10 Jan 2017 10:18:50 -0400
> From: Joshua Colp 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] PJSIP status check at DB level
> (Realtime)
> Message-ID:
> <1484057930.4162258.843096561.6368d...@webmail.messagingengine.com
> >
> Content-Type: text/plain; charset="utf-8"
>
> On Tue, Jan 10, 2017, at 10:11 AM, Ahmed Munir wrote:
> > Hi,
> >
> > I would like to know how to check PJSIP status for endpoints at DB level
> > (realtime) just like in chan_sip? Like on chan_sip when sip extension
> > gets
> > register regseconds, ipaddr, fullcontact & many more parameters get
> > updated
> > at DB level.
> >
> > The version of asterisk I'm using is 13.8 cert 4
>
> Registered devices are added as contacts to the "ps_contacts" table.
> This is because there can be multiple of them. You would need to look in
> that table and make the association (based on AOR).
>
> --
> 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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Re: [asterisk-users] sip:p...@noname.com

2017-01-11 Thread Joshua Colp 
On Wed, Jan 11, 2017, at 11:09 AM, Thufir Hawat wrote:
> The SIP trace shows messages from what I took to be a suspicious 
> connection from sip:p...@noname.com so I added that IP address to IP 
> tables...but then anveo showed as unreachable so I removed that rule.
> 
> Yes, I'm running fail2ban.
> 
> What are these messages from sip:p...@noname.com?  The domain name alone 
> set off alarm bells for me.  (I was looking for my own registration 
> attempts when I turned on SIP debugging.)

It's just a keepalive from their end to determine that your SIP server
is still reachable and to keep open any NAT mappings so you are
reachable.

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[asterisk-users] hangup locked channels

2017-01-11 Thread Dov Bigio 
Hello,

When I run "core show channels verbose", I seen around 10 locked channels
that are lasting hundreds of hours (I haven't restarted Asterisk for more
than 7 weeks).

I try to hangup those channels using "hangup request ", but
nothing happens.

What could I do (besides restarting Asterisk)?

Thanks
Dov
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Re: [asterisk-users] sip show [general]?

2017-01-11 Thread Carlos Rojas 
Hi

You can do

sip show settings


On Jan 11, 2017 5:32 AM, "Thufir Hawat"  wrote:

> I appreciate that the console lets you see the details for a peer with
> "sip show peer foo".  Certainly, I can look in sip.conf to see the
> [general] context, but can I output those settings, and only those
> settings, to the console?
>
>
>
> thanks,
>
> Thufir
>
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Re: [asterisk-users] sip show [general]?

2017-01-11 Thread John Kiniston 
'sip show settings' may do what you want.

On Wed, Jan 11, 2017 at 3:32 AM, Thufir Hawat 
wrote:

> I appreciate that the console lets you see the details for a peer with
> "sip show peer foo".  Certainly, I can look in sip.conf to see the
> [general] context, but can I output those settings, and only those
> settings, to the console?
>
>
>
> thanks,
>
> Thufir
>
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>
> New to Asterisk? Start here:
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Re: [asterisk-users] Dial() from the console?

2017-01-11 Thread Thufir Hawat 



On Wed, 11 Jan 2017, Doug Lytle wrote:


On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote:



Can I dial directly from the asterisk console with the Dial() application?



console dial number@context



Thanks, that's much more intuitive :)


-Thufir

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[asterisk-users] sip:p...@noname.com

2017-01-11 Thread Thufir Hawat 
The SIP trace shows messages from what I took to be a suspicious 
connection from sip:p...@noname.com so I added that IP address to IP 
tables...but then anveo showed as unreachable so I removed that rule.


Yes, I'm running fail2ban.

What are these messages from sip:p...@noname.com?  The domain name alone 
set off alarm bells for me.  (I was looking for my own registration 
attempts when I turned on SIP debugging.)




SIP trace:

fqdn*CLI>
fqdn*CLI> sip set debug on
SIP Debugging enabled
fqdn*CLI>

<--- SIP read from UDP:67.212.84.21:5010 --->
OPTIONS sip:s...@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 67.212.84.21:5010;branch=0
From: sip:p...@noname.com;tag=uloc-5875e606-bf5-dea1e-52564b36-00fe47a3
To: sip:s...@xxx.xxx.xxx.xxx:5060
Call-ID: cb004ab7-97b14601-e7ade23@67.212.84.21
CSeq: 1 OPTIONS
Content-Length: 0

<->
--- (7 headers 0 lines) ---
Sending to 67.212.84.21:5010 (NAT)
Looking for s in default (domain xxx.xxx.xxx.xxx)

<--- Transmitting (NAT) to 67.212.84.21:5010 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
67.212.84.21:5010;branch=0;received=67.212.84.21;rport=5010

From: sip:p...@noname.com;tag=uloc-5875e606-bf5-dea1e-52564b36-00fe47a3
To: sip:s...@xxx.xxx.xxx.xxx:5060;tag=as5f595fce
Call-ID: cb004ab7-97b14601-e7ade23@67.212.84.21
CSeq: 1 OPTIONS
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE

Supported: replaces, timer
Contact: 
Accept: application/sdp
Content-Length: 0


<>
Scheduling destruction of SIP dialog 
'cb004ab7-97b14601-e7ade23@67.212.84.21' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog 'cb004ab7-90004601-06ade23@67.212.84.21' 
Method: OPTIONS

Reliably Transmitting (NAT) to 67.212.84.21:5010:
OPTIONS sip:sip.anveo.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK601302be;rport
Max-Forwards: 70
From: "asterisk" ;tag=as194a0afc
To: 
Contact: 
Call-ID: 6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4
Date: Wed, 11 Jan 2017 14:56:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE

Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:67.212.84.21:5010 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
xxx.xxx.xxx.xxx:5060;branch=z9hG4bK601302be;rport=5060;received=xxx.xxx.xxx.xxx

From: "asterisk" ;tag=as194a0afc
To: ;tag=a1766e4537c6d6082807422b1789bf43.b9ae
Call-ID: 6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060
CSeq: 102 OPTIONS
Server: Anv Edge Proxy 3.5
Content-Length: 0

<->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 
'6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060' Method: OPTIONS

fqdn*CLI> sip set debug off
SIP Debugging Disabled
fqdn*CLI>
fqdn*CLI> sip show peers
Name/username HostDyn 
Forcerport ComediaACL Port Status  Description
anveo/1234567890  67.212.84.21Yes 
Yes5010 OK (78 ms)
demo_alice(Unspecified)D  Yes 
Yes0UNKNOWN
demo_bob  (Unspecified)D  Yes 
Yes0UNKNOWN
piter (Unspecified)D  Yes 
Yes0UNKNOWN
thufir(Unspecified)D  Yes 
Yes0UNKNOWN
5 sip peers [Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 
offline]

fqdn*CLI>
fqdn*CLI> sip show peer anveo


  * Name   : anveo
  Description  :
  Secret   : 
  MD5Secret: 
  Remote Secret: 
  Context  : from-anveo
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : 
  Language :
  Tonezone : 
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Max forwards : 0
  Dynamic  : No
  Callerid : "" <>
  MaxCallBR: 384 kbps
  Expire   : -1
  Insecure : port,invite
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL  : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID: No
  Path support : No
  Path : N/A
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode : rfc2833
  Timer T1 : 500
  Timer B  : 32000
  ToHost   : sip.anveo.com
  Addr->IP : 67.212.84.21:5010
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP

Re: [asterisk-users] Dial() from the console?

2017-01-11 Thread Doug Lytle 
>>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote:

>>> Can I dial directly from the asterisk console with the Dial() application?


console dial number@context

Doug

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[asterisk-users] Dial() from the console?

2017-01-11 Thread Thufir Hawat 

Can I dial directly from the asterisk console with the Dial() application?


or, is channel originate preferred:

channel originate SIP/thufir extension 18003569377@outbound





thanks,

Thufir

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[asterisk-users] sip show [general]?

2017-01-11 Thread Thufir Hawat 
I appreciate that the console lets you see the details for a peer with 
"sip show peer foo".  Certainly, I can look in sip.conf to see the 
[general] context, but can I output those settings, and only those 
settings, to the console?




thanks,

Thufir

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