[asterisk-users] Issues

2006-12-05 Thread Arlen Nascimento
7-1 answered SIP/502-9823 ... -- Hungup 'Zap/7-1' This problems most happen with the fax and pos machines that are connected to a handytone 286. I've already look for these on google in some many ways, but i couldn't find nothing interesting. I'am using Asterisk CVS-v

[asterisk-users] Attended Transfer

2006-12-05 Thread Arlen Nascimento
But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to make it work? I've already looked at google so many times and nothing Does anybody have an idea?? Regards -- Arlen Nascimento ___ --Band

Re: [asterisk-users] Attended Transfer

2006-12-05 Thread Arlen Nascimento
to the > vacant digits on the dial pad (DTMF). With an ATA you would use FLASH (aka RECALL) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Rejecting a Call

2006-12-06 Thread Arlen Nascimento
se in the dial plan without answering a call? Cheers, Ray ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-us

Re: [asterisk-users] No music on hold?

2006-12-20 Thread Arlen Nascimento
Phil, did you add letter 'm' to your dial options?? exten => _XXX,1,Dial(SIP/XXX,60,m) Regards Arlen Nascimento On 12/20/06, Phil Finkler <[EMAIL PROTECTED]> wrote: I installed the asterisk-addons from source and installed them. It looks like it copied format_mp3.so

[asterisk-users] Peer doesn't answer

2012-01-15 Thread Arlen Nascimento
an't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts?

Re: [asterisk-users] Peer doesn't answer

2012-01-16 Thread Arlen Nascimento
_ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http

Re: [asterisk-users] Peer doesn't answer

2012-01-16 Thread Arlen Nascimento
basically CLI shows SIP/X called SIP/Y I answer the call on Y but X keeps ringing and then both hangup. On Mon, Jan 16, 2012 at 8:01 AM, Sammy Govind wrote: > Paste some SIP traces of the call while Unmonitored. > > > On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento <

Re: [asterisk-users] Peer doesn't answer

2012-01-16 Thread Arlen Nascimento
ink, but can't call to it. > When i turn off qualify, the status changes to unmonitored. In this case, > I can make calls in both directions but the call is never established. The > phone keeps ringing until 'ring time' expires even when I answer the call > on the phone/s

Re: [asterisk-users] Peer doesn't answer

2012-01-18 Thread Arlen Nascimento
wrote: > I'm only expecting NAT issues if not the latency issues. SIP traces of any > such calls will make more sense. > > > On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento < > arlen.nascime...@gmail.com> wrote: > >> the client is aware of the adverse enviro

Re: [asterisk-users] Peer doesn't answer

2012-01-18 Thread Arlen Nascimento
up you settled on to get decent voice > quality over the Sat link? Which codec are you using, and what is the > bandwidth usage?. Are you doing just one concurrent call, Or multiple?. > > - > Regards, > AJ Stanfield > > > ----- Original Message - > From: "Arle