[Asterisk-Users] DTMF tones not recognized...

Michael Manousos manousos at inaccessnetworks.com
Wed Mar 26 07:02:08 MST 2003


Carlos Crembil wrote:
> Hi guys,
> I've a strange problem.
> 
> My scenario is a linux box running asterisk, and a Cisco 800 in the same
> LAN. The system has been working fine, except for an old H323 driver i've
> compiled to asterisk. So, I've rebuilt the pwlib and openh323 libraries
> with a new version (a requisite for the new H323 driver), and I've compiled
> the H323 driver with the new source.
> 
> I didn't change my Cisco configuration. But after that, some strange thing
> happens: when I pick up my phone connected to Cisco, and dial my asterisk
> configured extension, Cisco connects fine to Asterisk, Asterisk answers and
> sends the welcome. But for any reason, it does not recongnize the DTMF
> tones I send with my phone.

CISCO probably sends DTMF inband. If this is the case then
the inband DTMF detection is done inside ASTERISK (dsp.c)
and not in the H.323 channel driver. I have a rather old
snapshot of ASTERISK source (~2 weeks old) and inband DTMF
detection works fine.

If the CISCO doesn't send DTMF inband then this is a
problem of the H.323 channel driver and I 'll have to
check it.

So, check to see how does you CISCO send DTMF.

Regards,
Michael.

> 
> There were no problems in the compilation stage of any module. I've been
> debugging, but I can't find where the problem is?
> 
> Has anyone suffer the same?
> 
> Regards,
> Carlos.
> 
> PWLib was v1.3.1, now is v1.4.11.
> Openh323 was v1.9.1, now is v1.11.7.
> H323 Support for Asterisk was v0.2, now is v0.5.1.
> Asterisk version is CVS-03/08/03-15:48
> 
> My oh323.conf file:
> ;------------------------------------------
> [general]
> listenAddress=0.0.0.0
> listenPort=1720
> connectPort=1720
> fastStart=yes
> h245Tunnelling=yes
> h245inSetup=yes
> inBandDTMF=yes
> silenceSuppression=no
> jitterMin=20
> jitterMax=60
> ipTos=none
> outboundMax=10
> inboundMax=10
> gatekeeper=DISCOVER
> userInputMode=TONE
> context=voip-h323
> 
> [register]
> alias=asterisk
> alias=123
> alias=0
> context=all-aliases
> alias=ASTERISK
> alias=666
> context=more-aliases
> alias=665
> context=all-prefixes
> gwprefix=00
> gwprefix=01
> context=more-stuff
> alias=664
> gwprefix=02
> 
> [codecs]
> codec=G711A
> frames=20
> ;------------------------------------------
> 
> And my extensions.conf file:
> ;------------------------------------------
> [demo]
> exten => s,1,Answer                     ; Answer the line
> exten => s,2,DigitTimeout,5             ; Set Digit Timeout to 5 seconds
> exten => s,3,ResponseTimeout,10         ; Set Response Timeout to 10
> seconds
> exten => s,4,SetMusicOnHold,default
> 
> exten => 21,1,Dial,OH323/21 at 192.168.50.253
> exten => 21,102,Voicemail,u21
> exten => 21,103,Goto(s,5)
> 
> exten => 22,1,Dial,OH323/22 at 192.168.50.253,10
> exten => 22,2,Goto(s,5)
> 
> exten => 31,1,Dial,OH323/31 at 192.168.50.79
> 
> [voip-oh323]
> include => demo
> ;------------------------------------------
> 
> Carlos Crembil
> Servicios Profesionales
> http://openware.biz
> eMail: ccrembil at openware.biz
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users





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