[asterisk-users] DTMF over IAX trunk ignoring last digit

2013-08-09 Thread Marcelo Terres
Hello.

Scenario: 9 servers connectec to each other over IAX trunks. Users
used to call to remote extensions and remote conferences (meetme) via
IAX.

Problem: all extensions from one server (just one) when try to attend
remote conferences had problems with PIN validation. If they use their
local conferences the problem doesn't occur.

All servers use same OS (Ubuntu 10.04 64 bits), same Asterisk version
(1.8.16.0) and same configurations.

DTMF type is rfc2833 and jitterbuffer is off. Codec is gsm and
trunk=yes in iax.conf.

I did some tests with read cmd and discover that If user press 4
digits, only 3 digits are recognized. If user press 5 digits, only 4
are recognized and so on. Looks like the last digit is always ignored.

Any ideas?

Regards,

-- 
Marcelo H. Terres
Propus Informática LTDA
http://www.propus.com.br

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] DTMF over IAX

2006-11-02 Thread Leo Ann Boon

Jason Walker wrote:
Ok sorry for not being specific.  I am having a problem when people 
outside call in to my number which terminates at VoicePluse then The 
send IAX to me and I do not get any tones. People press buttons but it 
just goes to the next dialplan fall through.  It happens 60-70% of the 
time.

extentions.conf
I'd suggest you capture the IAX packets to determine if voicepulse is 
actually sending you the DTMF digits out-of-band. DTMF in IAX is always 
OOB, so you should receive an IAX packet with the DTMF. If you're not 
seeing those packets, then the problem lies upstream.


Leo

___
--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] DTMF over IAX

2006-11-01 Thread Jason Walker
Ok sorry for not being specific.  I am having a problem when people 
outside call in to my number which terminates at VoicePluse then The 
send IAX to me and I do not get any tones. People press buttons but it 
just goes to the next dialplan fall through.  It happens 60-70% of the time.

extentions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

;OEM
exten = _12125551212,1,Goto(OEM,s,1)

[OEM]
exten = s,1,Answer()
exten = s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)})
exten = s,n,Background(Outsource)
exten = s,n,WaitExten(10)
exten = s,n,Goto(inside,133,1)
exten = 9,1,Background(OEM_Menu)
exten = 9,n,WaitExten(10)
exten = 9,n,Goto(0,1)
exten = 0,1,Goto(inside,133,1)

IAX.conf
[general]
jitterbuffer=yes
forcejitterbuffer=no
maxjitterbuffer=500
autokill=yes

   ; -
   ; IAX INCOMING USER
   ;
   ; This is the user for incoming calls from:
   ; connect02.voicepulse.com
   ; -
  
[voicepulse]   ; -- Name must be [voicepulse]

context=voicepulse-in  ; -- Should match the context you
  ; are using in extensions.conf
  ; to handle incoming calls
type=user
host=connect02.voicepulse.com
qualify=yes
notransfer=yes
disallow=all
allow=g729   ; -- List supported codecs
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=g726
allow=adpcm
allow=lpc10

___
--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] DTMF over IAX

2006-11-01 Thread Andrew Joakimsen
The problem is voicepulse, but they refuse to accept responsibility. From What phone are you pressing the DTMF?On 11/1/06, Jason Walker 
[EMAIL PROTECTED] wrote:Ok sorry for not being specific.I am having a problem when people
outside call in to my number which terminates at VoicePluse then Thesend IAX to me and I do not get any tones. People press buttons but itjust goes to the next dialplan fall through.It happens 60-70% of the time.
 extentions.conf[general]static=yeswriteprotect=noautofallthrough=yesclearglobalvars=nopriorityjumping=no;OEMexten = _12125551212,1,Goto(OEM,s,1)[OEM]exten = s,1,Answer()
exten = s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)})exten = s,n,Background(Outsource)exten = s,n,WaitExten(10)exten = s,n,Goto(inside,133,1)exten = 9,1,Background(OEM_Menu)
exten = 9,n,WaitExten(10)exten = 9,n,Goto(0,1)exten = 0,1,Goto(inside,133,1)IAX.conf[general]jitterbuffer=yesforcejitterbuffer=nomaxjitterbuffer=500autokill=yes; -
; IAX INCOMING USER;; This is the user for incoming calls from:; connect02.voicepulse.com; -
[voicepulse] ; -- Name must be [voicepulse]context=voicepulse-in; -- Should match the context you ; are using in extensions.conf ; to handle incoming calls
type=userhost=connect02.voicepulse.comqualify=yesnotransfer=yesdisallow=allallow=g729 ; -- List supported codecsallow=ulawallow=alaw
allow=gsmallow=ilbcallow=g726allow=adpcmallow=lpc10___--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing listTo 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] DTMF over IAX

2006-11-01 Thread Eric \ManxPower\ Wieling

Jason Walker wrote:
Ok sorry for not being specific.  I am having a problem when people 
outside call in to my number which terminates at VoicePluse then The 
send IAX to me and I do not get any tones. People press buttons but it 
just goes to the next dialplan fall through.  It happens 60-70% of the 
time.


DTMF issues like this are caused by a problem where the PSTN call is 
converted to IAX.  There is nothing you can do about it unless you 
manage the box that converts PSTN to IAX.

___
--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