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@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?
Thanks,
James Taylor
MetroTel
903-793-1956
--
Using Opera's revolutionary e-mail client: http://www.opera.com/m2/
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@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
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
Hi,
Someone have running a MTNT,SIP and Asterisk please let me know really I
don't know which way to take.
Greetings,
JC.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julio
Cesar Pinto
Sent: Wednesday, November 09, 2005 3:56 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] MAX TNT SIP / Asterisk
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
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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@lists.digium.com wrote:> Message: 8 > Date: Thu, 10 Nov 2005 13:19:20 -0500 > From: "Julio Cesar Pinto" <jc@ifxcorp.com> > Subject: RE: [Asterisk-Users] MAX TNT SIP / Asterisk > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <A60450238C4B1341AC25ECC027D8895801389964@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.-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051110/f864fb56/attachment.htm
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jeremiah
Millay
Sent: Thursday, November 10, 2005 3:55 PM
To: asterisk-users@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@lists.digium.com
wrote:
Message: 8
Date: Thu, 10 Nov 2005 13:19:20 -0500
From: "Julio Cesar Pinto" <jc@ifxcorp.com>
Subject: RE: [Asterisk-Users] MAX TNT SIP / Asterisk
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<A60450238C4B1341AC25ECC027D8895801389964@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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