[Asterisk-Users] MAX TNT SIP / Asterisk

[Julio Cesar Pinto]

Hi,

I'm implementing MAX TNT in SIP mode with Asterisk, and I couldn't
establish the connection.

So, reviewing the messages posted in this list I found a message with
date Nov 10 2004, a year ago :)

Well, I have the same problem posted by James Taylor; my configuration
is the same that Darren Bentley propose.

I'd like to know if some have more information about that.

Thanks in advance,

JC.

---
PDTA: I page the history.

Using Software version 10.1.0

Here's what I did:

1. Create a Media Profile (called "voip")

name* = voip
active = yes
protocol-type = sip

[in MEDIA-GATEWAY/voip:voip-options]
packet-audio-mode = g711-ulaw
frames-per-packet = 2
silence-det-cng = no
ena-adap-jitter-buffer = yes
max-jitter-buffer-size = 19
initial-jitter-buffer-size = 2
voice-ann-dir = /current
voice-ann-enc = g711-ulaw
call-inter-digit-timeout = 6000
silence-threshold = 0
dtmf-tone-passing = inband
maxcalls = 672
rfc2833-payload-type = 96
g711-transparent-data = no
rtp-problem-reporting = { no 30 60 }

[in MEDIA-GATEWAY/voip:sip-options]
t1-timer = 500
t2-timer = 4000
invite-retries = 6
non-invite-retries = 10
primary-proxy = { x.x.x.x "" 5060 compact } (IP ADDRESS OF ASTERISK)
secondary-proxy = { 0.0.0.0 "" 5060 compact }
registration-proxy = { x.x.x.x "" 5060 compact 1 } (IP ADDRESS OF
ASTERISK)
proxy-heartbeat = 0
proxy-failover-window = 60
reroute-on-proxy-failure = no
trusted-proxy =
unknown-ani = ""
blocked-ani = ""
privacy-proxy-require = disabled
cause-code-map = s
start-call-method = invite
trunk-group-options =
onhold-minutes = 0
support-100rel = disabled
internationalize = no
international-prefix = no
country-code = ""
national-destination-code = ""
local-number-ton = unknown-ton
call-transfer-method = ip-transfer
notify-timer = 0
invite-with-multiple-codecs = disabled

2. Configure Call Route for Digitam Modem card

admin> get call-route {{{1 3 0}0}0}
[in CALL-ROUTE/{ { { shelf-1 slot-3 0 } 0 } 0 }]
index* = { { { shelf-1 slot-3 0 } 0 } 0 }
active = yes
trunk-group = 0
phone-number = 7299 (last 4 digits of your DID)
preferred-source = { { any-shelf any-slot 0 } 0 }
call-route-type = voice-call-type
cost = 0

3. Configure the T1 ports

default-call-type = dnis-or-voip
media-gateway = voip

I did this about 8 months ago and don't have my notes with me so I hope
I remembered everything. Give it a shot. Good luck

- Darren

On Tue, 2004-11-09 at 09:49, Tim Connolly wrote:
> Do you have the TNT's config available? I'd love to see this work!
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Darren Bentley
> Sent: Monday, November 08, 2004 1:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] MAX TNT SIP / Asterisk
> 
> Have you attempted to use SIP? It's working quite well for me.
> 
> sip.conf
> 
> [maxtnt]
> type=friend
> host=xxx.xxx.xxx.xxx
> dtmfmode=inband
> callerid="MaxTNT" <maxtnt>
> context=toll-access
> qualify=yes
> reinvite=no
> canreinvite=no
> disallow=all
> allow=g729
> allow=ulaw
> 
> extensions.conf
> 
> (xxx.xxx.xxx.xxx would be the address of your MaxTNT)
> 
> [toll-trunks]
> ;
> ; Outbound 1-nxx-nxx-xxxx goes via: PSTN
> ;
> exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@xxx.xxx.xxx.xxx,60)
> exten => _1NXXNXXXXXX,2,Hangup
> 
> [local-trunks]
> ;
> ; Outbound to nxx-xxxx goes via: PSTN
> ;
> exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@xxx.xxx.xxx.xxx,60)
> exten => _NXXXXXX,2,Hangup
> ;
> 
> [local-access]
> ;
> ; Extensions that are this context are allowed to only call local PSTN
> numbers and other extensions
> ;
> include => extensions
> include => local-trunks         ; Access to Local numbers
> 
> [toll-access]
> ;
> ; Extensions that are this context are allowed to call local and long
> distance PSTN numbers and other extensions
> ;
> include => local-access         ; Everything local-access has
> include => toll-trunks          ; Access to toll numbers
> 
> - Darren
> 
> 
> On Mon, 2004-11-08 at 10:36, James Taylor wrote:
> > Your question indicates that there may be a better way...
> > ???
> > 
> > I want to use the voice mail and extension features of Asterisk, and

> > sometimes there is this NAT problem that Asterisk seems to handle
very  
> > well.
> > 
> > I've been using H.323 with the TNT.
> > 
> > 
> > Do you have an alternate solution?
> > 
> > 
> > On Mon, 8 Nov 2004 10:41:31 -0500 (EST), <alex at pilosoft.com>
wrote:
> > 
> > > On Tue, 2 Nov 2004, James Taylor wrote:
> > >
> > >> I can't get my MAX TNT to register with Asterisk.
> > >> TAOS 11.0.
> > >>
> > >> SIP phone registeration show up in Asterisk like this:
> > >>      <sip:user_name at ip_address> and works.
> > >>
> > >> The TNT shows up as:
> > >>      <sip:@ip_address>.
> > >>
> > >> Does anyone have this working?
> > >> Am I missing something here?
> > >> Where does the TNT get it's user name?  Or, can it work without
one?
> > > It works without one.
> > >
> > > Why do you need to register TNT to asterisk anyway?
> > >
> > > --alex