Yeah, I know, but when I last tried an upgrade to 1.4.18 it broke the whole IAX connectivity and I was forced to drop back.
I'll go: 1) Memory upgrade first 2) Clone the machine, and upgrade to latest 1.4.x However - my question would still stand, how exactly would I be able to debug whats going on in the RTP stream? And why its stuttering (sometimes halfway through a call). Any tips or tricks for actually debugging within Asterisk ? Thanks, Adrian -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: 02 June 2009 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: > Hi, > > It's a 2mb dedicated leased fibre line, with<50% utilisation. > My first thoughts were the internet link, but that wouldn't explain why > the client transmit (other channel), which is on the same LAN as the > server, would have the same problem at the same time. > > Gut feeling is that A*k was CPU overloaded, or the local LAN was, but > none of the stats show that going on at all, although my CPU stats are > 5min samples - so that might hide a 60s of intense CPU activity. > > It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. > Only runs Asterisk. > > Adrian > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve > Howes > Sent: 02 June 2009 14:23 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Call quality - how to debug > > > On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > >> Hi All, >> >> I've a 1.4.15 A*k server supporting several users (approx 80 total, >> but<10 sim calls usually). I've one user who complains of >> intermittent bad calls, though I suspect the bad calls are across >> the board, but intermittent. >> >> Inbound calls are via in IAX trunk from Gradwell. CPU stats say that >> Asterisk never uses more than 4-5% cpu, systems idle besides that. >> Memory seems ok too. Network utilisation is< 300kbps. The voice >> network (clients + server) sit on their own dedicated 100Mb >> switches. Stats from the switch say its lightly loaded. >> >> I've turned on voicefile recording. What we hear, when there is a >> bad call, is stuttered speech, from BOTH sides (so local SIP client, >> and remote IAX inbound call). >> Debug from asterisk just shows the call inbound, answered and then >> hung up as per normal. >> >> I'm at a loss of how to debug the voice issue further, without >> putting a wireshark PC on the switch, port-mirroring the server and >> then capturing all of the traffic in a round-robin-type capture and >> even then I'm not sure what that will achieve. >> >> I'm going to switch from IAX to SIP for the inbound calls for that >> user and see if that helps. >> >> Any ideas welcome, _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users