[asterisk-users] AW: asterisk-users Digest, Vol 27, Issue 23
Daniel Hikel
daniel at hikel.de
Thu Oct 5 07:10:54 MST 2006
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Today's Topics:
1. GXP - 2000 BLF (Andrew Shelton)
2. Re: AEL2 #include madness in Asterisk 1.4 - Murf? (Steve Murphy)
3. Re: Bandwidth requirements (Benny Amorsen)
4. RE: Extremely choppy sound on some of our POTSnetwork calls;
goes away with mute (sdgesa gaeharth)
5. Re: Re: Bandwidth requirements (J. Oquendo)
6. Re: snom 360: how to make record button working ? (Joe Pukepail)
7. Re: TNT Max Password reset (James)
8. two asterisk and one NBX system (jose diaz)
9. Re: Video Conference (Noah Miller)
10. RE: TNT Max Password reset (asterisk)
11. Re: TNT Max Password reset (Don)
12. Re: Re: extensions.conf strangeness (Michael Neuhauser)
----------------------------------------------------------------------
Message: 1
Date: Thu, 5 Oct 2006 14:00:41 +0100
From: "Andrew Shelton" <andrew.shelton at stemnetworks.co.uk>
Subject: [asterisk-users] GXP - 2000 BLF
To: <asterisk-users at lists.digium.com>
Message-ID:
<792989DB4816704C830932E501F2E79206BCCC at corp-svr-1.corporate.local>
Content-Type: text/plain; charset="us-ascii"
Hello,
I have been trying to get my Grandstream busy line filter to work for ages..
All the lights flash as they are supposed to.
If one Grandstream 7000 calls another Grandstream 7003 I can use Grandstream
7002 to pick the call up pressing the BLF button and all works fine.
However if I call Grandstream 7000 with a mobile phone and try to pickup the
call with Grandstream 7002 all I get is a 603 error on Grandstream 7002.
I am using firmware 1.1.12 for the Grandstream and 1.2.12.1 version of
asterisk
This is the error I get from my log..
if some one could please help
Oct 5 12:12:51 DEBUG[7723] chan_sip.c: build_route: Contact hop:
<sip:7003 at 192.168.1.94:5060>
Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing
NoOp("SIP/7003-b721be28", "**7002") in new stack
Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing
Pickup("SIP/7003-b721be28", "7002") in new stack Oct 5 12:12:51 DEBUG[8828]
app_directed_pickup.c: No originating channel found.
Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No call pickup
possible...
Oct 5 12:12:51 VERBOSE[8828] logger.c: == Spawn extension
(inbound-from-stem, **7002, 2) exited non-zero on 'SIP/7003-b721be28'
Oct 5 12:12:51 DEBUG[7716] channel.c: Avoiding initial deadlock for
'SIP/7003-b721be28'
SIP
[7000]
type=friend
context=inbound-from-stem
Subscribecontext=BLF
secret=*
host=dynamic
canreinvite=no
callgroup=2
pickupgroup=2
mailbox=7000 at default
username=7000
dtmfmode=rfc2833
callerid="STEM" <17524543545>
qualify=yes
EXTENSIONS
[default]
include => stem
include => to-siemens
include => BLF
include => BLF_group_pickup
[stem]
;exten STEM GROUP = 01752 692205
exten => 123454,1,Ringing
exten => 123454,n,Wait(1)
exten => 123454,n,Answer()
exten => 123454,n,NoOp(${CALLERID(all)})
exten => 123454,n,SetCIDName(Outside Caller)
exten => 123454,n,Set(CALLERID(number)=9${CALLERIDNUM})
exten => 123454,n,NoOp(${CALLERID(all)})
exten => 123454,n,Macro(stdexten2,7003,${STEMGROUP},20)
;exten 7000 = 01752 692204
exten => 123455,1,Ringing
exten => 123455,n,Wait(1)
exten => 123455,n,Answer()
exten => 123455,n,NoOp(${CALLERID(all)})
exten => 123455,n,SetCIDName(Outside Caller)
exten => 123455,n,Set(CALLERID(number)=9${CALLERIDNUM})
exten => 123455,n,NoOp(${CALLERID(all)})
exten => 123455,n,Macro(stdexten2,7000,${stem},20)
;exten 7001 = 01752 692283
exten => 123456,1,Ringing
exten => 123456,n,Wait(1)
exten => 123456,n,Answer()
exten => 123456,n,NoOp(${CALLERID(all)})
exten => 123456,n,SetCIDName(Outside Caller)
exten => 123456,n,Set(CALLERID(number)=9${CALLERIDNUM})
exten => 123456,n,NoOp(${CALLERID(all)})
exten => 123456,n,Macro(stdexten2,7001,${stem1},20)
[internal]
;Internal Extensions
exten => _7XXX,1,Ringing
exten => _7XXX,n,Wait(1)
exten => _7XXX,n,Answer()
exten => _7XXX,n,Set(FOO1=${CHANNEL:4})
exten => _7XXX,n,Set(FOO2=${CUT(FOO1,-,1)})
exten => _7XXX,n,Set(CALLERID(number)=${FOO2})
exten => _7XXX,n,Macro(stdexten,${EXTEN},SIP/${EXTEN})
[inbound-from-pstn] ; inbound calls to this context from outside lines
include => default
[inbound-from-sip]
include => default
[inbound-from-local]
;from sip default context used.. requires hints
include => voicemail
include => provider
include => outbound
;include => stem ;for hints
[inbound-from-stem]
include => BLF
include => internal
include => DefExt
include => voicemail
include => outbound
include => BLF_group_pickup
include => feature-cfu
include => feature-cfna
include => feature-cfb
[inbound-from-logicall]
include => internal
include => DefExt
include => voicemail
include => outbound
include => BLF_group_pickup
include => feature-cfu
include => feature-cfna
include => feature-cfb
;Test section for BLF on Grandstreams for Stem
[BLF_group_pickup]
include =>inbound-from-stem
exten => _**.,1,NoOp(${EXTEN})
exten => _**.,2,Pickup(${EXTEN:2})
exten => _**.,3,Hangup
[BLF]
include =>inbound-from-stem
exten =>7000,hint,SIP/7000
exten =>7000,1,Dial(SIP/7000,20,r)
exten =>7001,hint,SIP/7001
exten =>7001,1,Dial(SIP/7001,20,r)
exten =>7002,hint,SIP/7002
exten =>7002,1,Dial(SIP/7002,20,r)
exten =>7003,hint,SIP/7003
exten =>7003,1,Dial(SIP/7003,20,r)
exten =>7004,hint,SIP/7004
exten =>7004,1,Dial(SIP/7004,20,r)
exten =>7005,hint,SIP/7005
exten =>7005,1,Dial(SIP/7005,20,r)
exten =>7006,hint,SIP/7006
exten =>7006,1,Dial(SIP/7006,20,r)
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Message: 2
Date: Thu, 05 Oct 2006 07:10:00 -0600
From: Steve Murphy <murf at digium.com>
Subject: [asterisk-users] Re: AEL2 #include madness in Asterisk 1.4 -
Murf?
To: asterisk-users at lists.digium.com
Message-ID: <1160053800.3638.88.camel at monster>
Content-Type: text/plain; charset="us-ascii"
On Thu, 2006-10-05 at 01:08 -0700, dgarstang at oneeighty.com wrote:
> Asterisk 1.4 beta2.
>
> My top level /etc/asterisk/extensions.ael has the following
> two lines:
>
> #include "include/syst/extensions.ael"
> #include "include/btck/extensions.ael"
>
> Here is the console output on Asterisk load.
>
> app_system.so => (Generic System() application)
> [Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3798
> pbx_load_module: Starting AEL load process.
> [Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3805
> pbx_load_module: AEL load process: calculated config file name
> '/etc/asterisk/extensions.ael'.
> [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
> --Read in included
> file /etc/asterisk/include/syst/extensions.ael, 4130 chars
> [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
> --Read in included file /etc/asterisk/include/syst/macros.ael,
> 1463 chars
> [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
> --Read in included
> file /etc/asterisk/include/syst/dundiapps.ael, 758 chars
> [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
> --Read in included file /etc/asterisk/include/syst/rdapps.ael,
> 275 chars
> [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
> --Read in included
> file /etc/asterisk/include/btck/extensions.ael, 1385 chars
> [Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3808
> pbx_load_module: AEL load process: parsed config file name
> '/etc/asterisk/extensions.ael'.
> [Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto:
> Error: file /etc/asterisk/include/syst/extensions.ael, line
> 157-157: goto: no label remote exists in the current
> extension!
> [Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto:
> Error: file /etc/asterisk/include/syst/extensions.ael, line
> 159-159: goto: no label local exists in the current
> extension!
> [Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:3821 pbx_load_module:
> Sorry, but 0 syntax errors and 2 semantic errors were
> detected. It doesn't make sense to compile.
> pbx_ael.so => (Asterisk Extension Language Compiler)
>
> Here's the context
> from /etc/asterisk/include/syst/extensions.ael, that contains
> lines 157 that the parser is complaining about:
>
> 148 context syst_Route {
> 149
> 150 _[*0123456789]. => {
> 151 NoOp(*** Originated call ${CALLERID} ->
> ${EXTEN});
> 152 Set(TMP=${CALLERID(number)});
> 153 &SysLogger(This is a test message);
> 154 &FastAGIConnectGet(CALLERID);
> 155 ChanIsAvail(SIP/${EXTEN});
> 156 if ("${AVAILCHAN}" = "") {
> 157 goto remote;
> 158 } else {
> 159 goto local;
> 160 }
> 161 remote:
> 162 NoOp(REMOTE);
> 163 Set(PATH=
> ${DUNDILOOKUP(3254103,DUNDIRegistr)});
> 164 //Set(PATH=
> ${DUNDILOOKUP(${EXTEN},DUNDIRegistr)});
> 165 Dial(${PATH});
> 166 Hangup();
> 167 local:
> 168 NoOp(LOCAL);
> 169 Dial(SIP/${EXTEN});
> 170 Hangup();
> 171
> 172 }
> 173 }
>
> As you can quite clearly see, labels 'remote' and 'local' DO
> exist in the syst_Route context.
>
> Now, if I switcheroo the two includes around in the top
> level /etc/asterisk/extensions.ael, to:
>
> #include "include/btck/extensions.ael"
> #include "include/syst/extensions.ael"
>
> and reload Asterisk, I get:
>
> [Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3813
> pbx_load_module: AEL load process: compiled config file name
> '/etc/asterisk/extensions.ael'.
> [Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3816
> pbx_load_module: AEL load process: merged config file name
> '/etc/asterisk/extensions.ael'.
> [Oct 4 15:57:28] WARNING[1202]: pbx.c:6194
> ast_context_verify_includes: Context 'syst_PSTNStart' tries
> includes nonexistent context 'syst_AppACDQueue'
> [Oct 4 15:57:28] WARNING[1202]: pbx.c:6194
> ast_context_verify_includes: Context 'btck_CallStart' tries
> includes nonexistent context 'syst_ACD'
> [Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3819
> pbx_load_module: AEL load process: verified config file name
> '/etc/asterisk/extensions.ael'.
> pbx_ael.so => (Asterisk Extension Language Compiler)
>
> There are no errors about nonexistent labels in the syst_Route
> extension. I would not have thought that #include order made
> any difference, since all we are doing is pulling a bunch of
> contexts into a global context space.
>
> Anyone? Mr Murpy, care to take a shot at it? :)
>
> Doug.
Doug--
I cannot reproduce the problems, given just the one context.
There is something magical about your data, that the code trips over it, and
to find the bugs, I will need your files! Is this possible?
As to order, you are correct, it should not make a difference what order the
files are included in the data.
I did note that in the above output, you got the error messages:
[Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 ast_context_verify_includes:
Context 'syst_PSTNStart' tries includes nonexistent context
'syst_AppACDQueue'
[Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 ast_context_verify_includes:
Context 'btck_CallStart' tries includes nonexistent context 'syst_ACD'
These messages do not come from the AEL compiler, but rather, are complaints
from the bowels of the asterisk engine: somewhere, it's not finding some
included contexts... which may mean yet one more bug in the AEL code: why
didn't AEL make note of it first?
murf
--
Steve Murphy
Software Developer
Digium
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------------------------------
Message: 3
Date: 05 Oct 2006 15:31:54 +0200
From: Benny Amorsen <benny+usenet at amorsen.dk>
Subject: [asterisk-users] Re: Bandwidth requirements
To: asterisk-users at lists.digium.com
Message-ID: <m3lknuubw5.fsf at ursa.amorsen.dk>
Content-Type: text/plain; charset=us-ascii
>>>>> "rJ" == raphael Jacquot <sxpert at sxpert.org> writes:
rJ> ATM cell tax is actually 10% as there's 5 header bytes for each 53
rJ> bytes cell,
For VoIP the cell tax is much larger. In the example, each RTP packet
contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't fit in
one cell, so you end up with 106 bytes at the ATM layer to transport 20
bytes of G.729. The ATM-caused overhead is thus 46 bytes per voice packet,
thereby making the needed bandwidth 77% larger.
All in all VoIP over ADSL adds 430% overhead, when using G.729 and 20ms
packets. Lovely, isn't it?
/Benny
------------------------------
Message: 4
Date: Thu, 5 Oct 2006 06:38:10 -0700 (PDT)
From: sdgesa gaeharth <pollux1234567890 at yahoo.com>
Subject: RE: [asterisk-users] Extremely choppy sound on some of our
POTSnetwork calls; goes away with mute
To: asterisk-users at lists.digium.com
Message-ID: <20061005133811.67985.qmail at web50808.mail.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"
Below is the text of my original post. I am not sure what Codec we are
using. The "Codec Preferences" phone setting shows, in order of
preference, G.711u, G.711A, G.729AB
We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core
4-2.6.14-1.1656_FC4smp. It is installed on a Dell PE 2500 with 2x900 MHz
processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium TDM400P
card which is connected to 4 POTS lines. The server is also connected to a
100MB switched LAN where we have about 20 Polycom 501 phones with the
latest firmware updates. Nothing else runs on the server except an ftp
daemon which is never used except when a phone reboots.
For about 20% of the calls to the outside world, the voice on the other
end of an outside line is incredibly choppy. Enough to where we have to
hang up and call on a cell phone. It is always the same numbers that are
choppy. The funny thing is, if I press mute while talking on a choppy
call, the choppiness goes away completely.
I have tried: turning off ACPI, turning off APCI, moving the
card to another PCI slot, changing the RX/TX gains. There are no shared
IRQs. I have tested the lines by unplugging them from the asterisk server
and plugging them directly into an analogue phone. Using "cat
/proc/interrupts; sleep 10 ; cat /proc/interrupts" I see that there are
about 1,000 interrupts per seconds between the card and the CPU.
I do not think it is a network congestion problem as intra-office
communications as well as voicemail retrieval are always perfect. The Voip
does not go over any routers, just a max of 2 switches with a 1GB trunk.
This happens even off-hours when the network isnt being used at all.
There are never more than 2 people on the phone at the same
time and it is definitely not an over-utilized processor.
I have trying to figure this out for 2 months on and off with no
success any help is appreciated.
Thanks
Andrew Shelton <andrew.shelton at stemnetworks.co.uk> wrote: v\:*
{behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:*
{behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) } What codec are you
using?
How many phone? What load is the server under?
---------------------------------
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of sdgesa
gaeharth
Sent: 05 October 2006 13:22
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Extremely choppy sound on some of our
POTSnetwork calls; goes away with mute
1)Can anyone tell me how to do this on a Polycom 501?
2)Can you explain why you think this any why it ony happens on some calls?
Thanks
Andres <andres at telesip.net> wrote:
>
>
> For about 20% of the calls to the outside world, the voice on the
> other end of an outside line is incredibly choppy. Enough to where
> we have to hang up and call on a cell phone. It is always the same
> numbers that are choppy. The funny thing is, if I press mute while
> talking on a choppy call, the choppiness goes away completely.
>
>
>
Maybe you have silence suppression enabled on your phones. Try to
disable it and see if it helps.
>------------------------------------------------------------------------
>
>
>
--
Andres
Technical Support
http://www.telesip.net
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Message: 5
Date: Thu, 05 Oct 2006 09:38:07 -0400
From: "J. Oquendo" <sil at infiltrated.net>
Subject: Re: [asterisk-users] Re: Bandwidth requirements
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <45250ABF.8070506 at infiltrated.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Benny Amorsen wrote:
>>>>>> "rJ" == raphael Jacquot <sxpert at sxpert.org> writes:
>>>>>>
>
> rJ> ATM cell tax is actually 10% as there's 5 header bytes for each 53
> rJ> bytes cell,
>
> For VoIP the cell tax is much larger. In the example, each RTP packet
> contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't
> fit in one cell, so you end up with 106 bytes at the ATM layer to
> transport 20 bytes of G.729. The ATM-caused overhead is thus 46 bytes
> per voice packet, thereby making the needed bandwidth 77% larger.
>
>
CRTP solves this issue (40byte waste)
--
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net
The happiness of society is the end of government.
John Adams
------------------------------
Message: 6
Date: Thu, 5 Oct 2006 08:42:16 -0500
From: "Joe Pukepail" <pukepail at gmail.com>
Subject: Re: [asterisk-users] snom 360: how to make record button
working ?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<7b3aa3a40610050642l41c41980xa283758f0072e742 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
There was a patch to get this working, looks like it has been abandoned,
though. Should give you a starting point to get it working, or perhaps a
bounty would get someone interested in getting it usable and committed.
http://bugs.digium.com/view.php?id=4845
On 10/4/06, Joel Hill <jhill at asteriskit.com.au> wrote:
>
> Hi Noro,
>
> Depending on what firmware you have this is the way to go.
> Go to the Functions keys page, then look for the Record button, Change
> the type to DTMF and in number put in *1 which is the default Asterisk
> recording function.
>
> Hope this helps
>
> Cheers,
>
> Joel
> Asterisk IT
> www.asteriskit.com.au
>
>
> noro kamen wrote:
> > Hi,
> >
> > I'd like to make record button working on snom 320/360 + asterisk.
> >
> > As I learned from wireshark output, the phone produces SIP info
> > message "Record: on", while record button pressed.
> >
> > Can anybody give me an advice, how to teach asterisk to understand
> > that SIP info message and start recording ?
> >
> > TIA
> > noro
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
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Message: 7
Date: Thu, 5 Oct 2006 08:43:26 -0500
From: "James" <jltaylor at metrotel.net>
Subject: Re: [asterisk-users] TNT Max Password reset
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <014d01c6e884$3bd21a70$2c05a8c0 at table>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
I have five MAX TNT's runnig with SIP and g.729.
They will do E1's, T1's, T3's.
James Taylor
1-903-793-1956
----- Original Message -----
From: "Steve Kennedy" <steve-asterisk at gbnet.net>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, October 05, 2006 4:28 AM
Subject: Re: [asterisk-users] TNT Max Password reset
> On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:
>
>> On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
>> > Anyone have happen know how to reset the password on a TNT Max?
>> > Thanks.
>> Does your asking here suggest that the the MAX's can do, say, voice
>> gateway service? Protocols? Codecs?
>
> Ascent TNT's with the right software and hardware can do SIP, E1
> termination/origination, and all sorts of codecs.
>
> Similar functionality to Cisco AS5200'ish.
>
>
> Steve
>
> --
> NetTek Ltd UK mob +44-(0)7775 755503
> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
> Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve at gbnet.net
> Euro Tech News Blog http://eurotechnews.blogspot.com
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------------------------------
Message: 8
Date: Thu, 05 Oct 2006 09:50:54 -0400
From: jose diaz <ing.josediaz at verizon.net.do>
Subject: [asterisk-users] two asterisk and one NBX system
To: asterisk-users at lists.digium.com
Message-ID: <45250DBE.7020300 at verizon.net.do>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
We have three servers: Two asterisk and one NBX 3COM.
The connection between Asterisk1 and Asterisk2 is with IAX2.
The connection between Asterisk2 and NBX is with a Digium analog
TDM400P (2FXO and 2 FXS)
The dial plan Asterisk1: 3XXX
The dial plan Asterisk2: 2XXX
The dial plan NBX: 1XXX
The system work well, but the call from Asterisk1 to NBX fail. I'm using
the IAX2 protocol to call from asterisk1 to asterisk2, i need to
trasnfer the call to the NBX. How i can to make that?
Regards,
Jose Diaz
------------------------------
Message: 9
Date: Thu, 5 Oct 2006 10:01:43 -0400
From: "Noah Miller" <noahisaacmiller at gmail.com>
Subject: Re: [asterisk-users] Video Conference
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<8699dcab0610050701m32130387w56a2299b4965d702 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi Bilal -
> We need to apply Video conference, can asterisk
> support this?
No. Asterisk supports video calls between two end points, but not
video conferences with three or more participants.
There is a bounty for someone to add this feature, but nobody has
successfully implemented it yet.
> What I need for that?
Something else. You can get video conferencing software, or if you
have the right hardware you can use it. There are many hardware video
conferencing units available from Polycom, Tandberg, Sony, etc.
- Noah
------------------------------
Message: 10
Date: Thu, 5 Oct 2006 16:04:55 +0200
From: "asterisk" <asterisk at sirtem.fr>
Subject: RE: [asterisk-users] TNT Max Password reset
To: "'James'" <jltaylor at metrotel.net>, "'Asterisk Users Mailing List -
Non-Commercial Discussion'" <asterisk-users at lists.digium.com>
Message-ID: <200610051404.k95E4iku025944 at ra.sirtem.fr>
Content-Type: text/plain; charset="iso-8859-1"
Hello james,
I have 1 max with pri, only used for incomming data call.
It is a old box, where to find firmware for this unit ?
If a can use it for voice....
Ps: i leave in France..
Many thanks...
-----Message d'origine-----
De : asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] De la part de James
Envoyi : jeudi 5 octobre 2006 15:43
@ : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] TNT Max Password reset
I have five MAX TNT's runnig with SIP and g.729.
They will do E1's, T1's, T3's.
James Taylor
1-903-793-1956
----- Original Message -----
From: "Steve Kennedy" <steve-asterisk at gbnet.net>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, October 05, 2006 4:28 AM
Subject: Re: [asterisk-users] TNT Max Password reset
> On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:
>
>> On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
>> > Anyone have happen know how to reset the password on a TNT Max?
>> > Thanks.
>> Does your asking here suggest that the the MAX's can do, say, voice
>> gateway service? Protocols? Codecs?
>
> Ascent TNT's with the right software and hardware can do SIP, E1
> termination/origination, and all sorts of codecs.
>
> Similar functionality to Cisco AS5200'ish.
>
>
> Steve
>
> --
> NetTek Ltd UK mob +44-(0)7775 755503
> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
> Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve at gbnet.net
> Euro Tech News Blog http://eurotechnews.blogspot.com
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
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Message: 11
Date: Thu, 5 Oct 2006 10:04:11 -0400
From: "Don" <sales at xwebfactor.com>
Subject: Re: [asterisk-users] TNT Max Password reset
To: "James" <jltaylor at metrotel.net>, "Asterisk Users Mailing List -
Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Message-ID: <02e401c6e887$23506c60$1d01a8c0 at shizznit2000>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=response
We used to use em...
I believe you can just use a serial connection to them and reset them...
Could be mistaken been a couple years now...
----- Original Message -----
From: "James" <jltaylor at metrotel.net>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Thursday, October 05, 2006 9:43 AM
Subject: Re: [asterisk-users] TNT Max Password reset
>I have five MAX TNT's runnig with SIP and g.729.
> They will do E1's, T1's, T3's.
>
> James Taylor
> 1-903-793-1956
>
>
> ----- Original Message -----
> From: "Steve Kennedy" <steve-asterisk at gbnet.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, October 05, 2006 4:28 AM
> Subject: Re: [asterisk-users] TNT Max Password reset
>
>
>> On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:
>>
>>> On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
>>> > Anyone have happen know how to reset the password on a TNT Max?
>>> > Thanks.
>>> Does your asking here suggest that the the MAX's can do, say, voice
>>> gateway service? Protocols? Codecs?
>>
>> Ascent TNT's with the right software and hardware can do SIP, E1
>> termination/origination, and all sorts of codecs.
>>
>> Similar functionality to Cisco AS5200'ish.
>>
>>
>> Steve
>>
>> --
>> NetTek Ltd UK mob +44-(0)7775 755503
>> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
>> Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve at gbnet.net
>> Euro Tech News Blog http://eurotechnews.blogspot.com
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> _______________________________________________
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>
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.407 / Virus Database: 268.12.13/463 - Release Date: 10/4/2006
>
>
------------------------------
Message: 12
Date: Thu, 05 Oct 2006 16:07:14 +0200
From: Michael Neuhauser <mike at firmix.at>
Subject: Re: [asterisk-users] Re: extensions.conf strangeness
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Cc: Brian Candler <B.Candler at pobox.com>
Message-ID: <1160057234.9804.13.camel at willow.firmix.at>
Content-Type: text/plain; charset="us-ascii"
On Thu, 2006-10-05 at 11:12 +0100, Brian Candler wrote:
> Is there a debug mode which can say:
>
> "dialplan: trying to match 611 against pattern _1XXXXX: failed
> dialplan: trying to match 611 against pattern _2XXXXX: failed
> dialplan: trying to match 611 against pattern _6X.: matched"
No, there isn't (I assume to keep this central part as fast as possible,
i.e., even "if (option_debug) ..." costs time and pollutes the cache).
I've created and attached a one line patch (for 1.4 branch, r44464) that
should give you the info you need (sort of). But be aware that I haven't
tested it on 1.4 (only on 1.2, but things are different there). Only use
this patch on a test system as it will generate massive amounts of
output and will considerably slow down call handling.
--
Dr. Michael Neuhauser mailto:mike at firmix.at
Firmix Software GmbH sip:mike at firmix.at
Vienna/Austria/Europe tel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at/
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