Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?

[Thufir Hawat]

I would be very interested in using sipsak for something like this.  What 
have you tried so far?



-Thufir

On Mon, 16 Jan 2017, Olivier wrote:


Thinking over my previous, I wonder if sipsak could be used to send
outgoing SIP NOTIFY messages.
Would both Asterisk and sipsak be able to share networks resources ?

Thoughts ?

2017-01-16 14:10 GMT+01:00 Olivier :

[..]







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

[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

[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
  

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

[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]?

[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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[asterisk-users] anveo, a different kind of trunk provider?

[Thufir Hawat]

Anveo has their config as so:

[anveo]
type=friend
host=sip.anveo.com
port=5010
username= ACCOUNT_NUMBER
secret= SIP_PASSWORD
insecure=port,invite
disallow=all
allow=ulaw
context=from-anveo

http://www.anveo.com/faq.asp?code=sip_asterisk



but this seems slightly odd.  I have an account with them where my hard 
phone, an SPA 942 IP phone, connects directly to them.  I just entered the 
SIP details.  Presumably they're running Asterisk and have it configured 
for my SIP account.


But, their registration string with Asterisk is:

 Locate [general] secion and add the following
register => ACCOUNT_NUMBER:sip_passw...@sip.anveo.com:5010


Wouldn't this send every outbound call through that Anveo account?

Let's say that I add a more hard or softphones, but configure them to 
connect to my Asterisk server running on AWS.  When Anveo dials out to the 
POTS everything shows as coming from a single number, the account number?





thanks,

Thufir

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