Any chance this is a simple directmedia and/or NAT issue? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, October 03, 2014 4:14 PM To: tjrl...@live.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lost audio on forwarded calls
Asterisk does not need to care. Is it SIP all the way through? Thanks, Steve T On Fri, Oct 3, 2014 at 3:12 PM, Todd R. <tjrl...@live.com<mailto:tjrl...@live.com>> wrote: OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think that can relate :-) Here goes.. Inbound calls flow like this: Tier 1 Provider (SIP) > Asterisk 1.8 > Name Brand PBX - Calls work fine Outbound calls flow like this: Name Brand PBX > Asterisk 1.8 > Tier 1 provider (SIP) - Calls work fine Problem is being reported on that many (not all) calls have no audio when they are forwarded. Example of forwarded call: Inbound call comes in from Tier 1 Provider > Asterisk 1.8 > Name Brand PBX Name Brand PBX then forwards the call back out to users cell phone: Name Brand PBX > Asterisk 1.8 > Tier 1 provider No audio a large percentage of the time. It's my opinion that the Asterisk box only sees the forwarded call as a regular outbound call and forwards it on to the Tier 1 provider then to the users cell phone. I don't see how Asterisk even knows or cares if it was forwarded within the Name Brand PBX. The Name Brand PBX is the one making the connection of the inbound and outbound call. All other inbound and outbound calls are fine, audio is only lost when the Name Brand PBX connects the two calls and creates the forward. Thoughts? -- _____________________________________________________________________ -- 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
