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. 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@192.168.50.253 exten => 21,102,Voicemail,u21 exten => 21,103,Goto(s,5) exten => 22,1,Dial,OH323/22@192.168.50.253,10 exten => 22,2,Goto(s,5) exten => 31,1,Dial,OH323/31@192.168.50.79 [voip-oh323] include => demo ;------------------------------------------ Carlos Crembil Servicios Profesionales http://openware.biz eMail: ccrembil@openware.biz
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@192.168.50.253 > exten => 21,102,Voicemail,u21 > exten => 21,103,Goto(s,5) > > exten => 22,1,Dial,OH323/22@192.168.50.253,10 > exten => 22,2,Goto(s,5) > > exten => 31,1,Dial,OH323/31@192.168.50.79 > > [voip-oh323] > include => demo > ;------------------------------------------ > > Carlos Crembil > Servicios Profesionales > http://openware.biz > eMail: ccrembil@openware.biz > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, I'll clarify you the case.
Asterisk with the older versions of PWLib, Openh323, and H323 support
worked fine with my Cisco configuration (I've only rebuilt these libraries
and modules, and not Asterisk). Cisco's configuration is extremely easy:
just an FXS port configured with codec G711a to connect to asterisk when I
dial digit "0".
After the compilations, when I started asterisk again, the problem
appeared.
Carlos Crembil
Servicios Profesionales
http://openware.biz
eMail: ccrembil@openware.biz
Michael Manousos
<manousos@inaccessnetworks. Para:
asterisk-users@lists.digium.com
com> cc:
Enviado por: Asunto: Re:
[Asterisk-Users] DTMF tones not recognized...
asterisk-users-admin@lists.
digium.com
26/03/2003 11:02 a.m.
Por favor, responda a
asterisk-users
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.
> 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
>
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Thank you Michael. I've turned inbandDTMF to "no" and now is
working.
Regards,
Carlos.
Carlos Crembil
Servicios Profesionales
http://openware.biz
eMail: ccrembil@openware.biz
Michael Manousos
<manousos@inaccessnetworks. Para:
asterisk-users@lists.digium.com
com> cc:
Enviado por: Asunto: Re:
[Asterisk-Users] DTMF tones not recognized...
asterisk-users-admin@lists.
digium.com
26/03/2003 11:02 a.m.
Por favor, responda a
asterisk-users
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@192.168.50.253
> exten => 21,102,Voicemail,u21
> exten => 21,103,Goto(s,5)
>
> exten => 22,1,Dial,OH323/22@192.168.50.253,10
> exten => 22,2,Goto(s,5)
>
> exten => 31,1,Dial,OH323/31@192.168.50.79
>
> [voip-oh323]
> include => demo
> ;------------------------------------------
>
> Carlos Crembil
> Servicios Profesionales
> http://openware.biz
> eMail: ccrembil@openware.biz
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users