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