I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would really like to know what debug level to use (if any) that would allow me to see that the Phone Event codec packets are being relayed from the Cisco to the GW. Finally, if the GW was unable to convert phone events to DTMF tones, will Asterisk generate the tones on the GW call leg if I configure the SIP phone for out-of-band DTMF and the SIP GW for in-band? Thanks for your help!
I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would really like to know what debug level to use (if any) that would allow me to see that the Phone Event codec packets are being relayed from the Cisco to the GW. Finally, if the GW was unable to convert phone events to DTMF tones, will Asterisk generate the tones on the GW call leg if I configure the SIP phone for out-of-band DTMF and the SIP GW for in-band? Thanks for your help! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users