Re: [Asterisk-Users] Calculating required bandwidth
You can encapsulate it as ppp, still some overhead, but less I think than HDLC. Ed - Original Message - From: "Kevin P. Fleming" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, December 16, 2004 1:36 PM Subject: Re: [Asterisk-Users] Calculating required bandwidth > Andrew Kohlsmith wrote: > > > RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it > > is actually costlier: I'll spare you all the calculations but it's 20 > > channels of SIP G.711 audio per T1, likely with enough room for > > signalling. :-) > > And you can't run straight IP over a T1 circuit either; it's usually > framed in HDLC frames. There's a little more overhead for you > > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux base for small Asterisk server?
I'm using * in a prepaid environment with Whitebox Respin with all the bells and whistles loaded. I made the move when Redhat lost their mind and discontinued support of 9.0. The asterisk is still in test and development mode (not a lot of traffic yet) but it seems to work okay. I am, however running an older version of asterisk CVS from May 2004, as versions after that gave me fits with some of the apps I depend on (playinterruptibletones for instance). I doubt the problems I encountered were the result of the O/S, more likely they are the result of "improvements?" to the asterisk CVS that conflict with some of the apps I'm using. Installing, compiling and running * on Whitebox was straight forward and trouble free. - Original Message - From: "João Amaro" <[EMAIL PROTECTED]> To: "Bill Bradford" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, November 04, 2004 11:47 AM Subject: Re: [Asterisk-Users] Best Linux base for small Asterisk server? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi I'm using Fedora RC1 without problems (kernel 2.4.22) Anyone here have tried Whitebox Respin 1 with asterisk ? Maybe nest week i'll install a small Asterisk Server on it. Asterisk + Apache only. Bill Bradford wrote: | I'm in the process of building up a small (1x1) test Asterisk box | based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a | FX100P). | | Anyone have suggestions as to the best Linux distribution (or | kernel) to base the system on? | | I'll just have one FXO/POTS line and then a Grandstream Budgetone | 101 IP phone; this is more for playing with IVR functionality than | anything else. | | Thanks. | | Bill ___ Asterisk-Users | mailing list [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users To | UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Manuel João S. Costa Amaro <[EMAIL PROTECTED]> ICQ: 57398499 MSN: [EMAIL PROTECTED] "As únicas pessoas que aprecio são os loucos: os que são loucos para viver, loucos para falar, loucos para se salvar, desejosos de tudo ao mesmo tempo; os que nunca bocejam nem dizem lugares-comuns, mas que ardem, se inflamam e brilham como fabulosos fogos-de-artifício." (Jack Kerouac) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBimtIJUm/Bor63CERAsV+AJ9ns8COTviXq1wuq1ulRbAr017HywCgmQES EvLwOwbb64aZoNs0Lsg/PrY= =7olh -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Centrex
It shouldn't matter if the inbound line is centrex, it's just a phone line
ringing into asterisk at that point. For the outbound side of the equation,
you'll have to dial 9 (or other digit as defined by the system) get dialtone
and send the dialed digits.
You might try:
exten => _9XX,1,dial(zap/g3/${EXTEN})
replace zap/g3 with whatever you're using to dial outbound. This should put
you on the track to figuring out how to get centrex outbound working.
- Original Message -
From: "Tim Thompson" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Monday, November 01, 2004 3:13 PM
Subject: RE: [Asterisk-Users] Centrex
> Centrex is a type of line and I do not believe there is a compatible card
> for *.
>
> FXS isn't going to cut it. Centrex is a digital type line.
>
> Tim.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
> Sent: Monday, November 01, 2004 1:35 PM
> To: Asterisk Users
> Subject: [Asterisk-Users] Centrex
>
> OK folks,
> I'm trying to help get another remote Asterisk box up and running.
> The system currently has a single FXO card, but it doesn't seem to
> be working, my guess is because the inbound line is CENTREX. Knowing
> nothing about Centrex, can someone tell me if I'm right, and need an
> FXS card?
>
> Thanks,
> Tim
>
> --
> ><
> >> Tim Sailer >< Coastal Internet, Inc. <<
> >> Network and Systems Operations >< PO Box 726 <<
> >> http://www.buoy.com >< Moriches, NY 11955 <<
> >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 <<
> ><
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Re: [Asterisk-Users] mwi over serial port
I'd be interested in looking at the code. I work for a CLEC/ISP and have access to sundry DMS and virtually every possible circuit configuration. Testing and debugging most implementations would be pretty easy for me. - Original Message - From: "Clay Zevely" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Wednesday, October 13, 2004 10:54 AM Subject: RE: [Asterisk-Users] mwi over serial port > Can I get the code? > > Clay > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Kent > Claussen > Sent: Wednesday, October 13, 2004 10:40 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] mwi over serial port > > > > We have the SMDI interface running to a DMS 10. If anyone is interested let > us know. The code would need a little clean up to get released. > Kent > > > > On 10/13/04 10:29 AM, "Michael Welter" <[EMAIL PROTECTED]> wrote: > > > The bounty is bogus, the offerors are not serious, and they should take > > it off the wiki. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and software Raid
I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I can experiment with. I've been wanting to use Redhat with software Raid 1 on an Asterisk server. Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Thanks all
Re: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed
I work for a CLEC in Dallas where I'm presently utilizing an Option 11 and
*. If you want to send the config files, I'll be happy to take a look at
them and see if I can spot any inadequacies.
- Original Message -
From: "asterisk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, July 13, 2004 9:38 AM
Subject: RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed
> The config files would be great, thanks !
>
> I'll let you know how I get on :)
>
> Julian
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
> Sent: 13 July 2004 15:16
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed
>
> We've successfully integrated with an Option 61c, but it was painful.
> We've set up both ends to emulate a 5eSS switch. The Asterisk is using
> pri_net (meaning the Nortel is pri_cpe (Client Side)). Unfortunately, in
> this configuration the Nortel thinks that this trunk is connected to an
> external phone company, so it always sends it's external Caller ID to it.
> That means that when someone on the Nortel calls someone on the Asterisk,
> you will always see the external caller id, not the actual extension from
> which the call originated.
>
> Our company uses Qwest to administer the Nortel, so it was the Qwest
> technicians who actually installed the card, set up the trunk and
> established the dial plan. We also found that we had to buy a special
> software option call "Custom Dial Plan" (CDP)which cost an extra $6,795
> including installation. With CDP installed on the Nortel, they were able
to
> create a dial-plan where extensions from 4000 to 4999 were sent down the
> trunk that's connected to the Asterisk. Asterisk then routes them
> accordingly. Asterisk has a dial-plan where all extensions from 2000 to
> 2999 are sent back to the Nortel. In that case, caller id works as it
> should for both name and number. Asterisk is also configured to send toll
> calls through the Nortel and that works correctly.
>
> So, that's the summary of what we've been able to accomplish. I can
provide
> you with the config files on the Asterisk, but you'll need a Nortel tech
for
> the rest; I have neither the ability nor the access to make those types of
> changes to a Nortel system. I hope this helps.
>
> Joe
> [EMAIL PROTECTED]
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of asterisk
> Sent: Tuesday, July 13, 2004 7:11 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed
>
>
> I've tried to do it myself, but my head is now bleeding from hitting it on
> the wall so much.
>
> We need someone who knows asterisk and Meridian PRI cards to help! If
> required, we will pay for a day's consultancy in order to get this thing
> working.
>
> Or, do I need to scrap my plans to keep the meridian system (60 phones
> ...) ... Please say no .. :)
>
> Please contact me offline ("asterisk" at "dotr" dot "com") if you want to
> sell yourselves :)
>
> Julian.
>
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[Asterisk-Users] playinterruptibletones
Since I downloaded the latest * CVS, I can't get playinterruptibletones to work. I've installed the patch from bugs.digium.com and recompiled and I still do not have the playinterruptibletones command What happened? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_addon_mysql.c
Following the latest * CVS update, my MySQL was broken. Following the update, Asterisk-addons would compile fine, but when I ran asterisk I got the following error: ERROR[1202489024]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas I then tried using the patch (bug id 0001823) from bugs.digium.com, and found that Asterisk-addons would no longer compile, giving me the following errors: make clean ; make install rm -f *.so *.o .depend ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: parse error before '!' token cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:110: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:110: (Each undeclared identifier is reported only once cdr_addon_mysql.c:110: for each function it appears in.) cdr_addon_mysql.c:121: `mysql' undeclared (first use in this function) cdr_addon_mysql.c: In function `my_unload_module': cdr_addon_mysql.c:226: `mysql' undeclared (first use in this function) cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:380: `mysql' undeclared (first use in this function) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:422: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I then tried using and older version of cdr_addon_mysql.c, and it also would not compile, but gave me an entirely different set of errors: ]# make clean ; make install rm -f *.so *.o .depend ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [cdr_addon_mysql.o] Error 1 I'm stuck on an MySQL project until I can resolve this problem. I've even blown away my system (OS as well as asterisk) and reloaded everything fresh from CVS, and still no joy. Any suggestions would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mysql errors
Since updating * via CVS earlier this week, I've been having problems with cdr_mysql. Prior to that time my queries and cdr all worked fine. Now, even though my queries still work, I get the messages similar to this: ERROR[1211374384]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas-- Hungup 'Zap/74-1' Anyone have any suggestions what went wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-addons mysql
I just downloaded the latest *CVS onto a freshly installed Redhat 9 system, and I noticed that the compile of asterisk-addons fails as follows: # make clean ; make install rm -f *.so *.o .depend ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [cdr_addon_mysql.o] Error 1 any ideas what happened? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HOW-TO DIFF
How do I patch an * file if all I have is the .diff file?
[Asterisk-Users] Asterisk Database
I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded number. Any suggestions on how this can be done? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disa & #
I want to be able to use DISA without sending the # sign to terminate the password. Actually, I rather be able to use a "fixed field of X digits" (i.e. 228 for the password, where if a match is found, the field is terminated and the call processes). Our system has been configured to act as a switch, basically, it accept's inbound calls, verifies account authorization, selects an outbound trunk, and routes the calls, think "call forwarding" and you'll have the basic concept. Having to allways use the # plays havoc with some of the Automatic route selection used in various PBX's (and even some automated dialers have problems processing the #) that I'm trying to route calls from. Essentially, I want these customers to be able to route calls through our asterisk without having to incur additional programming charges from their vendors. Being able to send an authorization string (or better yet being able to use their existing authorization string) that isn't terminated by # would make it a piece of cake to convert these accounts to our service. Has anyone had any experience passing authorization to DISA that isn't terminated with #? Has anyone got a recommendation as to a better way to achieve the same goal?
Re: [Asterisk-Users] exit
Try typing an ! followed by the enter key at the CLI prompt amd see what happens. - Original Message - From: "Fran Boon" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, February 27, 2004 7:20 AM Subject: Re: [Asterisk-Users] exit > Greg Kedrovsky wrote: > >>You must have started asterisk with "asterisk -c" > > No, I started it with "asterisk" and had it running in the background. > > Suggest starting as 'safe_asterisk' > > asterisk -r > exit > > Always works for me... > > F > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dual Xeon
When compiling Asterisk for a dual XEON based system are there any caveats or "switches" that we need to be aware of?
[Asterisk-Users] dtmf recording record and playback
I want to be able to record dtmf digits and then play them back as a voice file (i.e. your enter your telephone number into a file, and asterisk reads it back later as a voice file). The application is similar to voicemail applications, but would need to be able to parse the digits (i.e. 33 would be parsed as "thrity three", rather than "three, three", etc...). Has anyone got an idea how this could be implemented.
Re: [Asterisk-Users] Re: DISA
Yes I have tried immediate = yes.
I do get dialtone immediately when I go off-hook or dial in, but then
Asterisk won't accept any further input whether dialing from the Norstar or
dialing on the T1 side. Essentially, I can't break dialtone.
- Original Message -
From: "Robert Hajime Lanning" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, February 06, 2004 1:45 PM
Subject: Re: [Asterisk-Users] Re: DISA
> What is your zapata.conf?
> Have you tried "imediate = yes"?
>
>
> > John and sundry others:
> >
> > First thanks for your help.
> >
> > You have succiently summed up the problem. I do not get dialtone fast
> > enough.
> >
> > The following is a test dialplan that I set up this morning after
recieveing
> > the many kind e-mails, It's very basic, but it does allow me to process
a
> > call to my phone extension, albeit I still don't get dialtone
immediately
> > when I select a line or dial into the asterisk system. (see embedded
notes
> > for details).
> >
> > [general]
> > static = yes
> > writeprotect = no
> > ;
> > [main2]
> > exten => 9,1,dial(zap/g2)
> > exten => _5012
> > ignorepat => 9
> > ;
> > [main1]
> > exten => s,1,DISA(2285750,main2)
> > exten => s,2,Hangup( )
> > ;
> > ;Notes on testing:
> > ;Circuit is a full T1 provided by my in house Nortel
> > ;SL1 to port 3 of my Digium T410p. It's identified
> > ;in zaptel.conf as span =3,0,0,d4,ami., and configured
> > ;in zapata.conf as group=2, signalling=em_w,
> > ;channel = 49-72.
> > ;
> > ;For purposes of testing only, I have my Nortel Norstar
> > ;system with a T1 cartridge attached to port 4 of the
> > ;Digium T410p. It's identified in zaptel.conf as
> > ;span=4,0,0,esf,b8zs and configured in zapata.conf as
> > ;group=3, signalling=em_w, channel = 73-96.
> > ;
> > ;ztcfg -vv indicates the configuration is correct, and
> > ;zttool indicates that there are no errors
> > ;
> > ;When I select line 1 on the Norstar (where I would
> > ;normally expect to to get dialtone, in effect simply
> > ; going off hook) . I do not get dialtone.
> > ;
> > ;CLI indicates "Starting simple switch on 'Zap-73-1' ".
> > ;The same hold true if I dial in on this T1.
> > ;
> > ;after 5 seconds (the timeout), I finally recieve dialtone.
> > ;
> > ;At this point I dial "2285750#" and I get dialtone again
> > ;
> > ; CLI indicates "WARNING [1225991448]:
> > ;app_disa_c:290 : disa_exec: DISA on Zap/73-1
> > ;password is good".
> > ;
> > ;The dialplan then branches to [main2]
> > ;
> > [main2]
> > exten => 9,1,dial(zap/g2)
> > exten => _5012,1,dial(zap/g2)
> > ignorepat => 9
> > ;
> > ;Since both the Norstar and the SL1 are configured with
> > ;dial 9 access (and yes, I've tried using straight access
> > ;with the same results). I dial 995012, and the call
> > ;processes, ringing my extension 5012 on the SL1.
> > ;
> > ;CLI indicates
> > ;'Executing dial("Zap/73-1 , Zap/g2) in new stack'.
> > ;Called g2
> > ;'Zap/49-1 answered Zap/73-1'
> > ;'attempting native bridge of Zap/73-1 and Zap/49-1'
> > ;
> > ;I answer the call on my extension '5012' and talk as long
> > ;as I care and then simply hangup.
> > ;
> > ;CLI indicates 'Hungup 'Zap/49-1'
> > ;'spawn extension (main2,9,1) exited non-zero on
> > ;Zap/73-1'
> > ;Hungup 'Zap/73-1'
> > ;
> > [default]
> > exten => s,1,answer
> > exten => s,2,disa(no-password, main2)
> > exten => s,3,Hangup
> > ;
>
> --
> END OF LINE
>-MCP
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Re: [Asterisk-Users] Re: DISA
John and sundry others:
First thanks for your help.
You have succiently summed up the problem. I do not get dialtone fast
enough.
The following is a test dialplan that I set up this morning after recieveing
the many kind e-mails, It's very basic, but it does allow me to process a
call to my phone extension, albeit I still don't get dialtone immediately
when I select a line or dial into the asterisk system. (see embedded notes
for details).
[general]
static = yes
writeprotect = no
;
[main2]
exten => 9,1,dial(zap/g2)
exten => _5012
ignorepat => 9
;
[main1]
exten => s,1,DISA(2285750,main2)
exten => s,2,Hangup( )
;
;Notes on testing:
;Circuit is a full T1 provided by my in house Nortel
;SL1 to port 3 of my Digium T410p. It's identified
;in zaptel.conf as span =3,0,0,d4,ami., and configured
;in zapata.conf as group=2, signalling=em_w,
;channel = 49-72.
;
;For purposes of testing only, I have my Nortel Norstar
;system with a T1 cartridge attached to port 4 of the
;Digium T410p. It's identified in zaptel.conf as
;span=4,0,0,esf,b8zs and configured in zapata.conf as
;group=3, signalling=em_w, channel = 73-96.
;
;ztcfg -vv indicates the configuration is correct, and
;zttool indicates that there are no errors
;
;When I select line 1 on the Norstar (where I would
;normally expect to to get dialtone, in effect simply
; going off hook) . I do not get dialtone.
;
;CLI indicates "Starting simple switch on 'Zap-73-1' ".
;The same hold true if I dial in on this T1.
;
;after 5 seconds (the timeout), I finally recieve dialtone.
;
;At this point I dial "2285750#" and I get dialtone again
;
; CLI indicates "WARNING [1225991448]:
;app_disa_c:290 : disa_exec: DISA on Zap/73-1
;password is good".
;
;The dialplan then branches to [main2]
;
[main2]
exten => 9,1,dial(zap/g2)
exten => _5012,1,dial(zap/g2)
ignorepat => 9
;
;Since both the Norstar and the SL1 are configured with
;dial 9 access (and yes, I've tried using straight access
;with the same results). I dial 995012, and the call
;processes, ringing my extension 5012 on the SL1.
;
;CLI indicates
;'Executing dial("Zap/73-1 , Zap/g2) in new stack'.
;Called g2
;'Zap/49-1 answered Zap/73-1'
;'attempting native bridge of Zap/73-1 and Zap/49-1'
;
;I answer the call on my extension '5012' and talk as long
;as I care and then simply hangup.
;
;CLI indicates 'Hungup 'Zap/49-1'
;'spawn extension (main2,9,1) exited non-zero on
;Zap/73-1'
;Hungup 'Zap/73-1'
;
[default]
exten => s,1,answer
exten => s,2,disa(no-password, main2)
exten => s,3,Hangup
;
- Original Message -
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, February 05, 2004 9:55 PM
Subject: Re: [Asterisk-Users] Re: DISA
> At 9:32 PM -0500 2/5/04, Steve Creel wrote:
> >On Thu, 5 Feb 2004, John Todd wrote:
> >
> >>So, to boil your problem down to what I think is the problem:
> >>
> >>"When you attach an inbound call to the DISA application, it does not
> > >produce a dialtone fast enough."
> >
> >
> >
> >>>[main1]
> >>>;
> >>>; Take any number, and give it to the DISA. The DISA
> >>>; just then takes anything typed in within the (unchangeable)
> >>>; timer values, and hands it off to main2 to be post-processed.
> >>>; I include the standard i,h,t values for pedantic reasons.
> >>>;
> >>>exten => _X.,1,DISA(no-password,main2)
> >>>exten => _X.,2,Hangup
> >>>;
> >>>exten => h,1,Hangup
> >>>exten => i,1,Congestion
> >>>exten => i,2,Hangup
> >>>exten => t,1,Congestion
> >>>exten => t,2,Hangup
> >
> >
> >Not to point out the obvious, but isn't the delay he's seeing caused by
> >the _X. and the digittimeout? Couldn't this be resolved by using a more
> >specific match on the DISA instead of _X. ?
> >
> >Steve
> >[EMAIL PROTECTED]
>
> Ah, yes, that's probably the case. Without further information from
> the poster about how he was getting calls into the context, I assumed
> that this was a PRI or something that handed a DID to the context.
> If this is an FXO or some type of T1 trunking, then yes, the "s"
> extension would be more appropriate if this was an "immediate=yes"
> type of situation.
>
> GIGO.
>
> JT
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