[Asterisk-Users] Re: DISA SPA3000 issues
Thanks for the feedback, but the route that that I'm finding doesn't work is: Asterisk - SPA3000 - ZAP/BRI - Asterisk - DISA The problem appears to be on outbound calls from the SPA3000 where the second dial tone seems to stop audio transmission, changing the DTMF method make no difference. :( Thanks Dave Hawkes Philippe Lindheimer wrote: Just tried it on mine, worked fine: Cellphone Call - POTS - SPA3000 - Asterisk - DISA - Telasip As an FYI, I have my SPA3000 setup with INFO for the DTMF. When I originally installed it, I couldn't get the DTMF digits to work coming in using AUTO, which is why I have it using INFO (needs to be set on both the SPA and in Asterisk). I'm running the SPA300 with Software Version: 2.0.13(GWg), Hardware Version 2.0.1(4e16). philippe From: Dave Hawkes [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 17 May 2006 13:44:43 -0400 Subject: [Asterisk-Users] Re: DISA SPA3000 issues I have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug? Dave Hawkes Alchaemist wrote: Hi, These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was true no RTP DTMF events (RFC2833) are sent after the SPA listens the second dialtone. I just reviewed the 87 pages PDF of the SPA3000... and didn't find anything about such feature. Now I am going to try to figure out if it has something to do with the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on something I found on this forum, without the dialtone. Did anyone find out anything about this issue before? REGARDS!!! Alchaemist ___ --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 Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone. Get Yahoo! Messenger with Voice http://us.rd.yahoo.com/mail_us/taglines/postman10/*http://us.rd.yahoo.com/evt=39663/*http://messenger.yahoo.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 ___ --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
[Asterisk-Users] Re: DISA SPA3000 issues
I have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug? Dave Hawkes Alchaemist wrote: Hi, These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was true no RTP DTMF events (RFC2833) are sent after the SPA listens the second dialtone. I just reviewed the 87 pages PDF of the SPA3000... and didn't find anything about such feature. Now I am going to try to figure out if it has something to do with the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on something I found on this forum, without the dialtone. Did anyone find out anything about this issue before? REGARDS!!! Alchaemist ___ --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 ___ --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
[Asterisk-Users] Re: DISA SPA3000 issues
Just tried it on mine, worked fine:Cellphone Call - POTS - SPA3000 - Asterisk - DISA - TelasipAs an FYI, I have my SPA3000 setup with INFO for the DTMF. When I originally installed it, I couldn't get the DTMF digits to work coming in using AUTO, which is why I have it using INFO (needs to be set on both the SPA and in Asterisk).I'm running the SPA300 with Software Version: 2.0.13(GWg), Hardware Version 2.0.1(4e16).philippeFrom: Dave Hawkes [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 17 May 2006 13:44:43 -0400Subject: [Asterisk-Users] Re: DISA SPA3000 issuesI have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug?Dave HawkesAlchaemist wrote: Hi, These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was true no RTP DTMF events (RFC2833) are sent after the SPA listens the second dialtone. I just reviewed the 87 pages PDF of the SPA3000... and didn't find anything about such "feature". Now I am going to try to figure out if it has something to do with the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on something I found on this forum, without the dialtone. Did anyone find out anything about this issue before? REGARDS!!! Alchaemist ___ --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 Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone. Get Yahoo! Messenger with Voice___ --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
Re: [Asterisk-Users] Re: DISA SPA3000 issues
INFO is the way to go for DTMF at least on the PSTN tab of your SPA3K I have dtmfmode=auto in sip.conf I use DISA daily On 5/17/06, Philippe Lindheimer [EMAIL PROTECTED] wrote: Just tried it on mine, worked fine: Cellphone Call - POTS - SPA3000 - Asterisk - DISA - Telasip As an FYI, I have my SPA3000 setup with INFO for the DTMF. When I originally installed it, I couldn't get the DTMF digits to work coming in using AUTO, which is why I have it using INFO (needs to be set on both the SPA and in Asterisk). I'm running the SPA300 with Software Version: 2.0.13(GWg), Hardware Version 2.0.1(4e16). philippe From: Dave Hawkes [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 17 May 2006 13:44:43 -0400 Subject: [Asterisk-Users] Re: DISA SPA3000 issues I have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug? Dave Hawkes Alchaemist wrote: Hi, These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was true no RTP DTMF events (RFC2833) are sent after the SPA listens the second dialtone. I just reviewed the 87 pages PDF of the SPA3000... and didn't find anything about such feature. Now I am going to try to figure out if it has something to do with the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on something I found on this forum, without the dialtone. Did anyone find out anything about this issue before? REGARDS!!! Alchaemist ___ --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 ___ --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
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. snip [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 ___ 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] Re: DISA
What is your zapata.conf? Have you tried imediate = yes? quote who=Ed Devine 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 ___ 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] 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? quote who=Ed Devine 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 ___ 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] Re: DISA
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.
Is that the summary that I understand from your comments below? If
so, then we have narrowed things down a bit.
To the end of your actual problem, if I interpret it correctly: for
experimentation, try adding an Answer command right before the DISA
and see what you get. My experience with DISA (on VoIP and PRI, at
least) is that it gives an _immediate_ dialtone, without entry of any
keys or delay.
The unchangeable timer is the delay between the last keystroke of
entering something within the DISA and the DISA deciding to act upon
the string. As to my usage of the word unchangeable: this is open
source. Everything is changeable. My comments referenced what can
be done within the dialplan.
PS: Your mail program is confusing who said what, as well. I did not
say everything you attribute to me, and that is not clear from
looking at your message. James Sharp wrote the first two paragraphs
you say that are my quotes.
JT
At 7:43 PM -0600 2/5/04, Ed Devine wrote:
Andrew,
Thanks for your interest and courteous response. My company is a facilities
based CLEC. By way of background, I'm new to Asterisk and Digium, but I have
a good deal of past experience with Dialogic and NMS products in the telco
environment. I spend most of my time working with Nortel DMS-XXX switchgear
and managing the company ISP facilities.
I've been using a variation of a dialplan that I got from John Todd (see
below). The problem I've allways encountered is that for Asterisk to work in
our environment, it must allow the following:
Scenario:
user (whether automatic dialer, PBX ARS, PBX LCR, or even manually dialing
from any phone) accesses the switch (asterisk system) via a 10 digit did
number.
dial 972-NXX-
the switch answers and returns dialtone immediately
dial 228 1XX
(we use a seven digit authorization code sometimes in conjunction with
caller-id to verify that this is a valid account, etc...) followed
immediately by the 10 or 11 digit number you want to reach. The switch
selects an outbound trunk, strips the MSD if necessary, and ships the dialed
number digits.
The problem I've encountered is that inbound disa calls don't return
dialtone unless you enter something or until the unchangeable (John's word,
not mine) timer values time out.
John Todd has been most helpful, and his brief communications have been
incredible insights into how Asterisk works, he recently sent the following:
John's stuff starts here
app_disa will give answer and give you dial tone, wait for an
authentication code, then dump you into a context where you can make your
outgoing calls. Unfortunately, it needs a # at the end of the
authentication code.
A quick glance at the code suggests that it could be changed to expect a
fixed 7 digit access code.
It would be easy enough just to cut the first seven digits off the number
and run it through a comparison pass, and not use the authentication
routines at all.
;
; for North American numbers...
;
[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
;
;
[main2]
;
; Now, set the AUTHCODE to be the first seven digits of EXTEN
;
exten = _XXX1XX,1,SetVar(AUTHCODE=${EXTEN:0:7})
;
; ...and then forward this call out to a new context and extension,
; where the new extension is the 7th through 17th digit of the old EXTEN,
; which should translate into 1-123-456-7890 or whatever it was that
; the user entered as the desired destination phone number.
;
exten = _XXX1XX,2,Goto(main-dial-routine,${EXTEN:7:17},1)
;
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup
exten = t,1,Congestion
exten = t,2,Hangup
;
; end of example
This would end up (if the user entered the appropriate 7 digits and
1-npa-xxx- phone number) with passing the authentication code to
the main-dial-routine contained in ${AUTHCODE} and the ${EXTEN} set
to the number dialed.
You could also use the Cut application to perform a similar purpose
to my example using substring identifiers, if you wanted to put a
pound or star (or for those telephonically exotic among you, the
A/B/C/D) key separator in between the passcode digits and the phone
number.
JT
The upshot of my attempts was that, unless something is entered, dialtone
takes 5 seconds.
The same effect is apparent whether dialing inbound via the 10 digit did, or
when selecting a line from the Norstar
attached to the Digium T410P. If I dial in, the asterisk won't provide
dialtone unless I enter
Re: [Asterisk-Users] Re: DISA
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. snip [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] ___ 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] Re: DISA
quote who=Steve Creel [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. ? I think that would be right. I would have used: exten = s,1,DISA(no-password,main2) exten = s,2,Hangup -- END OF LINE -MCP ___ 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] 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. snip [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 ___ 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] Re: DISA and authcodes (was: t410p)
[moved from -dev, as the thread is better suited for -users]
At 5:10 PM -0600 1/30/04, James Sharp wrote:
I've pretty much got the routing covered at this point, I'm just not sure
how to get the Asterisk system to answer and give me dialtone immediately.
Any ideas or recommendations would be greatly appreciated.
app_disa will give answer and give you dial tone, wait for an
authentication code, then dump you into a context where you can make your
outgoing calls. Unfortunately, it needs a # at the end of the
authentication code.
A quick glance at the code suggests that it could be changed to expect a
fixed 7 digit access code.
It would be easy enough just to cut the first seven digits off the
number and run it through a comparison pass, and not use the
authentication routines at all.
;
; for North American numbers...
;
[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
;
;
[main2]
;
; Now, set the AUTHCODE to be the first seven digits of EXTEN
;
exten = _XXX1XX,1,SetVar(AUTHCODE=${EXTEN:0:7})
;
; ...and then forward this call out to a new context and extension,
; where the new extension is the 7th through 17th digit of the old EXTEN,
; which should translate into 1-123-456-7890 or whatever it was that
; the user entered as the desired destination phone number.
;
exten = _XXX1XX,2,Goto(main-dial-routine,${EXTEN:7:17},1)
;
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup
exten = t,1,Congestion
exten = t,2,Hangup
;
; end of example
This would end up (if the user entered the appropriate 7 digits and
1-npa-xxx- phone number) with passing the authentication code to
the main-dial-routine contained in ${AUTHCODE} and the ${EXTEN} set
to the number dialed.
You could also use the Cut application to perform a similar purpose
to my example using substring identifiers, if you wanted to put a
pound or star (or for those telephonically exotic among you, the
A/B/C/D) key separator in between the passcode digits and the phone
number.
JT
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