Hi Paul, No cigar unfortunately. I also tried encoding the message as gsm, ulaw and alaw with no success.
The ISDN interface is alaw and the SIP phones I was testing with are definately alaw. Not sure what to do from here. I might just need to bypass the issue using some alternate way to put the message in front of the inbound dialplan logic on some condition. aF On 01/09/2010, at 8:06 AM, Paul Belanger wrote: > On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara <[email protected]> wrote: >> Hi Paul, >> >> I tried adding Progress() to no avail. I still get no audio and below is >> what comes up in the console. >> > Try moving Progress() before the Dial(). If you Answer() the channel, > do you have the same problem? > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: [email protected] | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
