[Asterisk-Users] Re: MAX TNT SIP / Asterisk

Julio Cesar Pinto jc at ifxcorp.com
Thu Nov 10 16:48:59 MST 2005


Jeremiah, 
 
I'm glad to see that someone have working this schema, really I followed
the steps mentioned in the voip-info wiki, but without luck.
 
I see that the TNT is registered by Asterisk
 
*CLI> sip show peers
Name/username    Host            Dyn Nat ACL Mask             Port
Status   
maxtnt           10.0.43.2                 255.255.255.255  5060     OK
(16 ms)
 
The TNT have TAOS version 11.0.2
 
My config is the following, I appreciate is you help me see is a have a
wrong value.
 
TNT.
 
new MEDIA-GATEWAY
set name = voip
set active = yes
set protocol-type = sip
set voip-options packet-audio-mode = g711-ulaw
set sip-options primary-proxy ip-address = 10.0.43.4
set sip-options registration-proxy ip-address = 10.0.43.4
set sip-options registration-proxy register-interval = 1
write -f
 
new E1
set name = 1-2-1
set physical-address shelf = shelf-1
set physical-address slot = slot-2
set physical-address item-number = 1
set line-interface enabled = yes
set line-interface frame-type = 2ds
set line-interface signaling-mode = e1-mexico-signaling
set line-interface default-call-type = dnis-or-voice
set line-interface switch-type = switch-cas
set line-interface channel-config 1 channel-usage = switched-channel
set line-interface channel-config 17 channel-usage = switched-channel
set line-interface number-complete = 4-digits
set line-interface group-b-answer-signal = signal-b-1
set line-interface caller-id = get-caller-id
set line-interface collect-incoming-digits = yes
set line-interface media-gateway = voip
write -f
 
new DNIS
set dialed-number = 8812
write -f
 
new CALL-ROUTE
set index device-address physical-address slot = slot-4
set phone-number = 8812
set call-route-type = voice-call-type
write -f
 
 
extension.conf
[toll-trunks]
;
; Outbound 1-nxx-nxx-xxxx goes via: PSTN
;
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@10.0.43.2,60)
exten => _1NXXNXXXXXX,2,Hangup
 
[local-trunks]
;
; Outbound to nxx-xxxx goes via: PSTN
;
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@10.0.43.2,60)
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
 
sip.conf
 
[maxtnt]
type=friend
host=10.0.43.2
dtmfmode=inband
callerid="MaxTNT" <maxtnt>
context= toll-access
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=g729
allow=ulaw
 
 
I really appreciate if you send me your config to compare what I'm doing
wrong.
 
Greetings,
 
JC.
 
  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeremiah
Millay
Sent: Thursday, November 10, 2005 3:55 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: MAX TNT SIP / Asterisk
 
We are successfully using Lucent MAX TNT with Asterisk. Config is
essentially the same as the one found on voip-info wiki. Just do a
google on asterisk lucent tnt, and it should be one of the first pages
to pop up. We run our PRIs into the TNT, then talk SIP from the TNT to
our asterisk server. 
Jeremiah
 
 
On Nov 10, 2005, at 1:20 PM, asterisk-users-request at lists.digium.com
wrote:



Message: 8
Date: Thu, 10 Nov 2005 13:19:20 -0500
From: "Julio Cesar Pinto" <jc at ifxcorp.com>
Subject: RE: [Asterisk-Users] MAX TNT SIP / Asterisk
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
                <asterisk-users at lists.digium.com>
Message-ID:
 
<A60450238C4B1341AC25ECC027D8895801389964 at mailsrv.ifxcorp.com>
Content-Type: text/plain;     charset="us-ascii"
 
Hi,
 
Someone have running a MTNT,SIP and Asterisk please let me know really I
don't know which way to take.
 
Greetings,
 
JC.
 
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