[Asterisk-Users] Dial Tone + EM
Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail correctly (Dialogic D240-SC-T1) - without issues. I guess you recognize these are NOT PRI T1's - but old style DS1. However, when the external voice mail system begins to dial out, it grabs the port waits for the Wink and expects dial tone to be returned afterwards - Hearing none, it just sits there until the time out and gives up. My thinking is there should be an EM signaling type that CAN provide dial tone. - A quick scan of the source (chan_zap.c), it appears there is no such provisions for DT for any of the EM types. To me it appears to be a simple patch, but I'm sure I would screw it up if I attempt this myself, not being a programmer. And if by chance I would get it working, the next update would also need that patch. I'm hoping I can find someone on the list that is willing to add a new EM method with a DT provision and make it available to the release sources Thanks Bart = Zaptel.conf # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24; = seems like my only choice (em) # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,0,0,d4,ami em=25-48 ; = seems like my only choice (em) Zapata.conf: ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This is attached to CUST 3 VMS System ; signalling =em_w ; = might be wrong choice (see below for others) context=default group = 1 channel = 1-24 ; Span 2: TE4/0/3 TE410P (PCI) Card 0 Span 3 ; This T1 is WorldCom Local 714 DID's ; signalling =em_w ; = might be wrong choice (see below for others) context=from-did group = 3 channel = 25-48 Anybody have a clue for me TIA Bart ___ --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] Dial Tone
Hi, I try to generate a dial tone (tone you hear when you pick up the hook). The tone should be stopped as soon the user dials a single digit. Unfortunately Playtones(dial) don't stop until another extension is completely dialed. DISA doesn't work either with our Siemens Phones. The scenario looks like this: User wants to call the number 12345 1. User picks up the hook 2. User dials 0 - hears dial tone 3. User dials 1 - dial tone stops 4. User dials 2345 - phone 12345 is ringing Is there any solution for this? Regards, Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Dial Tone
On Wednesday 27 April 2005 12:12, Henry Jensen wrote:
Hi,
User wants to call the number 12345
1. User picks up the hook
2. User dials 0 - hears dial tone
3. User dials 1 - dial tone stops
4. User dials 2345 - phone 12345 is ringing
We're using chan_capi and had this same problem... The following really hacky
solution works OK with Asterisk 1.0.7, but not with CVS - I don't know why :)
[default]
exten = _120.,1,Goto(s,1) ; fax extensions are 1201 - 1208
exten = s,1,NoOp( incoming call from ISDN )
exten = s,2,Answer
exten = s,3,PlayTones(dial); Give the caller a familiar noise.
exten = s,4,DigitTimeout(0.1)
exten = s,5,WaitExten(0.1)
; next section captures the next digit and stops the dialtone
exten = _X,1,NoOp( Got a digit! It was ${EXTEN})
exten = _X,2,StopPlaytones()
exten = _X,3,SetVar(Predigits=${EXTEN}) ; Put that digit aside for
use later...
exten = _X,4,Goto(s-gathermoredigits,1)
exten = s-gathermoredigits,1,NoOp( Now looking for the rest of the number)
exten = s-gathermoredigits,2,DigitTimeout,3
exten = s-gathermoredigits,3,WaitExten(8) ; and give the caller 8
seconds overall to do their thing
; log + dial the composite number of Predigits + the remainder
exten = _X.,1,NoOp(${TIMESTAMP} ok, now we're going to dial
${Predigits}${EXTEN})
exten = _X.,2,Goto(outbound,${Predigits}${EXTEN},1)
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
The [outbound] context is jsut full of the normal exten = _01.,1,Dial(blaaah)
call routing
If someone has a better way of doing this, I'd be interested to hear it!
Cheers,
Gavin.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAL tone
Hey group! Could someone could help me configure a DIal plan in order that when i dial 9 at the beginning i receive DIAL TONE? Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone.___ 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] DIAL tone
This is in the notes in the default extensions.conf - ignorepat = 9 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gonzalo Gasca Meza Sent: Sunday, September 26, 2004 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] DIAL tone Hey group! Could someone could help me configure a DIal plan in order that when i dial 9 at the beginning i receive DIAL TONE? Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. ___ 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
