I am sending a call down a E&M wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=>799179,1,Dial(zap/g2,20,D(9179)) exten=>799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered. This means the digits have already been sent by the time the ringing phone is answered. Does anyone have an idea on how to signal this correctly? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071205/e2fe2977/attachment.htm
Jordan Novak wrote:> I am sending a call down a E&M wink trunk to a adtran tsu600 > channelbank. The extension is setup like so... > exten=>799179,1,Dial(zap/g2,20,D(9179)) > exten=>799179,2,Hangup() > It should Dial the Adtran and send some DTMF signals to a telephone on > an fxs module in the Adtran. > Asterisk is seeing the call answered when the T-1 is picked up by the > Adtran not when the ringing phone is answered. This means the digits > have already been sent by the time the ringing phone is answered. Does > anyone have an idea on how to signal this correctly?SendDTMF maybe? http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF Thanks, Steve Totaro
Where bandwidth is not an issue but good call quality is, is there any theoretical quality improvement to be had by using "slin" as the codec over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US). Does anyone know what the slin bandwidth is (compared to 64 kbps a-law). Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071205/fc522e26/attachment.htm
Partially answering my own question, it looks like "slin" is a 128 kbps codec. Peter ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Whisker, Peter Sent: 05 December 2007 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Use of "slin" as a codec Where bandwidth is not an issue but good call quality is, is there any theoretical quality improvement to be had by using "slin" as the codec over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US). Does anyone know what the slin bandwidth is (compared to 64 kbps a-law). Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071205/715aea3b/attachment.htm