login | register
Wed 01 of Aug, 2007 [09:20 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.48s
  • Memory usage: 2.22MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 2.43

Lucent TNT

Lucent Max TNT is used in large deployments as it supports multiple T1/E1 interfaces.
Takes 10U worth of rackspace!

It Can talk to Asterisk via both H.323 & SIP

Here is the TNT config to talk SIP:

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


Here is the sip.conf for Asterisk:

[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

Here is the extensions.conf for Asterisk:
(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



kentec says:

output of 'show' command:
Shelf 1 ( standalone ):
                                Reqd  Oper   Slot Type
   { shelf-1 slot-1 0 }         UP    UP     ether3-card
   { shelf-1 slot-2 0 }         UP    UP     8t1-card
   { shelf-1 slot-3 0 }         UP    UP     madd2-card

obfuscation in config:
netmask /29 (255.255.255.248)
10.0.0.1 - ip gateway
10.0.0.2 - asterisk
10.0.0.3 - maxtnt

10.0.0.11 - syslog host
10.0.0.12 - ntp host
10.0.0.13 - dns host
10.0.0.14 - dns host

There are "notes" inline with the config. This is a 'save' style
config, so can be copy/pasted straight in to your console (after
adjusting anything obfuscated and adjusting for card slots)


; start config
; we use g711-ulaw, rfc2833 (rtp) dtmf
new MEDIA-GATEWAY
set name = voip
set active = yes
set protocol-type = sip
set mg-sig-address ip-address = 10.0.0.3
set mg-rtp-address ip-address = 10.0.0.3
set transport-options type = udp
set voip-options codec-options g711-ulaw dtmf-tone-passing = rtp
set voip-options codec-options g711-ulaw silence-det-cng = yes
set sip-options primary-proxy ip-address = 10.0.0.2
set sip-options registration-proxy ip-address = 10.0.0.2
set sip-options unknown-ani = 0000000000
set sip-options unknown-name = www.kci.net
set sip-options blocked-ani = 0000000000
set sip-options blocked-name = blocked
set sip-options trunk-group-options otg-tag = ""
set sip-options trunk-group-options otg-tag-in-userinfo = no
set sip-options trunk-group-options dtg-tag = ""
write -f
;
; saving profiles of type SSH-SERVER-CONFIG
new SSH-SERVER-CONFIG
set ssh-enabled = yes
set login-grace-time = 60
write -f
;
; saving profiles of type CALL-ROUTE
; note this is the '8t1-card'
; { { { shelf-1 slot-2 0 } 0 } 0 }
new CALL-ROUTE
set index device-address physical-address slot = slot-2
set call-route-type = trunk-call-type
write -f
;
; note this is the 'madd2-card'
; note i disabled all call-routes but
; index 0 (voice-call-type)
; index 5 (voip-call-type)
; { { { shelf-1 slot-3 0 } 0 } 0 }
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set call-route-type = voice-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 1
set active = no
set call-route-type = phs-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 2
set active = no
set call-route-type = wormarq-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 3
set active = no
set call-route-type = digital-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 4
set active = no
set call-route-type = v110-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 5
set call-route-type = voip-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 6
set active = no
set call-route-type = rtfax-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 7
set active = no
set call-route-type = g729-call-type
write -f
;
new CALL-ROUTE
set index device-address physical-address slot = slot-3
set index entry-number = 8
set active = no
set call-route-type = mtp-link-type
write -f
;
; saving profiles of type LOG
new LOG
set call-info = end-of-call
set syslog-enabled = yes
set host = 10.0.0.11
set syslog-format = max
write -f
;
; saving profiles of type T1
; we only get channelized t1 from our local telephone monopoly (LTM)
; esf/b8zs trunk, eight channels
; asterisk switchtype 'national' so maxtnt 'switch-type = nat-isdn-2-pri'
; one thing that caused us problems was that we would see the call ring
; across several channels but not get answered by the max. turns out
; our LTM has an older switch and wasn't catching the default short
; 200ms wink. 'robbed-bit-mode = inc-w-400' and joy for all.
; only first eight channels active, the rest 'channel-usage = unused-channel'
; LTM DTMF's the incoming did so 'collect-incoming-digits = yes'
; set 'voip-gain-control output-pad = 9db-loss' because it was way
; too hot out, causing echo and some funky artifacting
; set 'egress-ani-dnis-format = dnis' because LTM cries if we send the
; extra DTMF for ANI on some NXX
new T1
set name = trunkone
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 = esf
set line-interface encoding = b8zs
set line-interface robbed-bit-mode = inc-w-400
set line-interface default-call-type = dnis-or-voip
set line-interface switch-type = nat-isdn-2-pri
set line-interface channel-config 1 trunk-group = 0
set line-interface channel-config 2 trunk-group = 0
set line-interface channel-config 3 trunk-group = 0
set line-interface channel-config 4 trunk-group = 0
set line-interface channel-config 5 trunk-group = 0
set line-interface channel-config 6 trunk-group = 0
set line-interface channel-config 7 trunk-group = 0
set line-interface channel-config 8 trunk-group = 0
set line-interface channel-config 9 channel-usage = unused-channel
set line-interface channel-config 10 channel-usage = unused-channel
set line-interface channel-config 11 channel-usage = unused-channel
set line-interface channel-config 12 channel-usage = unused-channel
set line-interface channel-config 13 channel-usage = unused-channel
set line-interface channel-config 14 channel-usage = unused-channel
set line-interface channel-config 15 channel-usage = unused-channel
set line-interface channel-config 16 channel-usage = unused-channel
set line-interface channel-config 17 channel-usage = unused-channel
set line-interface channel-config 18 channel-usage = unused-channel
set line-interface channel-config 19 channel-usage = unused-channel
set line-interface channel-config 20 channel-usage = unused-channel
set line-interface channel-config 21 channel-usage = unused-channel
set line-interface channel-config 22 channel-usage = unused-channel
set line-interface channel-config 23 channel-usage = unused-channel
set line-interface channel-config 24 channel-usage = unused-channel
set line-interface collect-incoming-digits = yes
set line-interface voip-gain-control output-pad = 9db-loss
set line-interface media-gateway = voip
set line-interface egress-ani-dnis-format = dnis
write -f
;
; saving profiles of type IP-ROUTE
new IP-ROUTE
set name = default
set gateway-address = 10.0.0.1
set metric = 1
set private-route = yes
write -f
;
; saving profiles of type IP-GLOBAL
new IP-GLOBAL
set domain-name = kci.net
set dns-primary-server = 10.0.0.13
set dns-secondary-server = 10.0.0.14
set system-ip-addr = 10.0.0.3
set sntp-info enabled = sntp-enabled
set sntp-info GMT-offset = utc-0700
set sntp-info host 1 = 10.0.0.12
write -f
;
new IP-INTERFACE
set interface-address physical-address shelf = shelf-1
set interface-address physical-address slot = slot-1
set interface-address physical-address item-number = 1
set ip-address = 10.0.0.3/29
set netmask = 255.255.255.248
write -f

-- kentec (running with this configuration for a month now with no issues)



See also:


Created by flavour, Last modification by kentec on Wed 01 of Jun, 2005 [20:15 UTC]

Comments Filter

Lucent TNT inbound call problem (h323)

by Mark frater on Tuesday 14 of February, 2006 [01:06:17 UTC]
I am trying to get DID calls from a TNT, using a Voicemaster ( think gnugk) as a H323 gatekeeper.
I think I have the Voicemaster set up ok as I can make calls to the same destination from other sources.

1)The TNT passes an ARQ to VM ok.
2) The GK passes an ACF back to the TNT ok
3) TNT fails the call and appears to drop the call. ie. stackH225Event: H.225 Idle release code = 267 (H323_Call_Network_Failed)
  A tcpdump verifies that NO packets are sent from the TNT to the destination Endpoint (202.89.130.49)

Can any TNT boffins out there give me some pointers?

The error codes below decode as follows:
269: H323_DROP_REASON_FEGW_CAUSE_CODE
267: H323_Call_Network_Failed

9014343.72: H323: -1: change state from IDLE --> to IDLE
9014343.72: H323: 127: _allocH323Call: inuse count = 1
9014343.80: H323: h323CallFromWhere(1204)
9014343.80: H323: h323InNewCall(1204): Incoming WAN call
9014343.80: H323: 127: dispatch msgType WAN_NEWCALL in state IDLE
9014343.80: H323: 127: _wanNewCall: SSD DNIS collected = 099771801
9014343.80: H323: 127: _wanNewCallRepeat()
9014343.80: H323: 127: _goToVoipSetupMode()
9014343.80: H323: 127: _goToVoipSetupMode: codec: 0
9014343.80: H323: 127: change state from IDLE --> to WAN_IN
9014343.80: H323: 127: _lanMakeCall: CLID collected = 93681030
9014343.81: H323: 127: _lanMakeCall: Tel. # = #099771801, stackID = 0
9014343.81: H323: 127: change state from WAN_IN --> to WAN_ORIG
9014343.81: New VOIP session name = V:eac99d51-a896-11da-9dcf-00d05
9014343.90: H323: 127: Received Admission Confirm
9014343.90: H323: 127: entering _handleVA(): call->state=WAN_ORIG, defaultMsg=(null pointer)
9014343.90: H323: 127: entering _handleVA(): non standard len is -1
9014343.90: H323: 127: stackH225Event: H323_LAN_INITIATED
9014343.95: H323: 127: stackH225Event: H.225 Idle release code = 267
9014343.95: H323: 127: stackH225Event: H323_LAN_DISCONNECTED
9014343.95: H323: 127: dispatch msgType LAN_DISCONNECTED in state WAN_ORIG
9014343.95: H323: 127: _lanCallDisconnected()
9014343.95: H323: 127: _toneOff()
9014343.95: H323: 127: _lanCallDisconnected() dropReasonWhy: 269
fegwCauseCode: 267
9014343.95: H323: 127: _goPlayCallFailurePrompt()
9014343.95: H323: 127: _goToBusyState(1204), dropFrom 4 dropReasonWhy=269 fegw=267
9014343.95: H323: 127: _goToVoipSetupMode()
9014343.95: H323: 127: _toneOn(6)
9014343.95: H323: 127: change state from WAN_ORIG --> to CALL_BUSY
Edit

Problem with registration

by Anonymous on Wednesday 22 of December, 2004 [18:48:18 UTC]
I have tried the config with a TNT running 11.x but it won't register with the asterisk box. Any ideas?


Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver