All;I was wondering if anyone has any experience for recording user voice while play background music?*My test case is :*When user enter in IVRS he is listen message for record your voice dial some digit after that user listen some background music ..
Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues.The card manages to grab a couple of (random) digits of the incoming CID, but theyre more or less useless.Is there any way to fix this?Aster..
Everyone,Just caught with another problem… When i got a voice mail in one of my account how would i email that voice message to email address. Any one have a..
The Asterisk Development Team has announced the release of Asterisk 10.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 10.8.0 resolves several issues reported by ..
The Asterisk Development Team has announced the release of Asterisk 184.108.40.206. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 220.127.116.11 resolves several issues reported..
I have had a case where after a hangup on the Alsa CHANNEL asterisk still thinks the line or call is active.I have:rtptimeout`rtpholdtimeout`rtpkeepalive`in my sip.conf file to help with this but it had no effect.How can I ensure a session HANGS up ..
EveryoneI have configure two sip turnk line on my trixbox now my client want If you dial turnk 1 DID and if that number is busy the call should automatically transfer to second line. Is there any way to do this.Please l..
We are experiencing an outage with at least issues.asterisk.org and potentially other services. We dont have an expectation of how long these services will be down and are currently in the process of troubleshooting.Digiums Asterisk Develo..
experts.Recently Ive insalled a PCI Khomp Pane on my server and inserted 4 chips to make call with it. The calls are good and no issue was noticed but I got reports that when someone call the chips the call volume is uncommonly low for both sides ..
I was wondering if anyone has any experience in streaming a MeetMe conference so that others might listen in to it?It would be nice if the audio format could be AAC, but at first any format will do.I did come across this: http://www.voip-info.org/wiki/index.p..