Re[2]: [Asterisk-Users] dtmf for public telephony access
Grazie Matteo, I looked in wiki pages, but found nothing regarding dtmf tone regeneration, just the indication that inbound tones are not allowed over low bitrate codecs. Would you raccomend sip info or rfc2833 as tone handling method ? P.S. finalmente un compatriota :) MB * hint : did you searched the ml first? MB this has been discussed a lot, even little time ago... MB however... MB sure, just use oob dtmf like rfc2833 or sip info dtmf... MB so you can use a low bitrate codec and asterisk MB will generate them again when going to the pstn... MB matteo MB Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto: Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ 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: Re[2]: [Asterisk-Users] dtmf for public telephony access
depends on the device you're using, if are supported or not. i feel very confortable with INFO method, since is a sip message and can be easily debugged :) Il gio, 2004-04-15 alle 09:45, Alessio Focardi ha scritto: Grazie Matteo, I looked in wiki pages, but found nothing regarding dtmf tone regeneration, just the indication that inbound tones are not allowed over low bitrate codecs. Would you raccomend sip info or rfc2833 as tone handling method ? P.S. finalmente un compatriota :) MB * hint : did you searched the ml first? MB this has been discussed a lot, even little time ago... MB however... MB sure, just use oob dtmf like rfc2833 or sip info dtmf... MB so you can use a low bitrate codec and asterisk MB will generate them again when going to the pstn... MB matteo MB Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto: Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ 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
[Asterisk-Users] dtmf for public telephony access
Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ 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] dtmf for public telephony access
* hint : did you searched the ml first? this has been discussed a lot, even little time ago... however... sure, just use oob dtmf like rfc2833 or sip info dtmf... so you can use a low bitrate codec and asterisk will generate them again when going to the pstn... matteo Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto: Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ 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
