[asterisk-users] destroying time

2012-06-15 Thread nakaji
? And why the time is about 30 sec ? Is this default time ? How to change tihis time ? any help appreciated. nakaji -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] destroying time

2012-06-15 Thread nakaji
thank you danny. I will try to write on sip.conf. regards nakaji - Original Message - From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 15, 2012 11:44 PM Subject: Re: [asterisk

[asterisk-users] When CALL-ID were same , I could hijack another session

2012-04-16 Thread nakaji
could hear the voice of A and C conference. Is this collect ?? Why I can hear another room's conference? Is this mean session hijack ?? I could do this. I want to know how to prevent this. any help appreciated. nakaji

[asterisk-users] meetme and jitterbuffer

2010-05-12 Thread nakaji
call, I have to change source code every time. It's only way. Is this right ? I think this is too inconvenient. thx. - nakaji -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] RTCP How to stop

2010-04-05 Thread nakaji
; Milliseconds between rtcp reports ;(min 500, max 6, default 5000) - nakaji Thank you for your infomation. It is regrettable that it is not easily revokable. There is no configuration option for doing this; RTCP is a mandatory part of an RTP implementation that intends to be compliant

Re: [asterisk-users] RTCP How to stop

2010-04-03 Thread nakaji
. -- - nakaji -- _ -- 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

[asterisk-users] RTCP How to stop

2010-04-02 Thread nakaji
Dear all; I want to stop RTCP from Asterisk-server to phone. But I want to use RTP. I looked rtp.conf/sip.conf, but I can't know about it. Please tell me how to stop RTCP only. Because , when I access under NAT, my gateway shutdown the port as gateway received RTCP from server. I use

Re: [asterisk-users] jitterbuffer and PLC

2010-01-20 Thread nakaji
I continued trying. Now I reached 2 results. 1. Asterisk ver1.6 or more has bug . When you want to use jitter and PLC and want to see packet-log , you will set ' jblog=yes ' on 'sip.conf '. But Asterisk can't make log-file. In /tmp/ packet-log-file will be made, if jb-modules work

Re: [asterisk-users] jitterbuffer and PLC

2010-01-18 Thread nakaji
Thank you for advice. Do you get the same results if you use: iax2 test losspct x Where x is the loss percent you'd like to test? Yes, I did it. On CLI show: VvvvLvvvLLvv vvLvvLvvv

Re: [asterisk-users] jitterbuffer and PLC

2010-01-18 Thread nakaji
hi. What user are you running Asterisk as? I tried 2 patarn. First , I worked asterisk as 'asterisk', and tested. But jitter and PLC didn't work correct. So I thought it may be caused permission problem, and made a new system working asterisk as 'root'. Now I tested as root. And same

[asterisk-users] jitterbuffer and PLC

2010-01-15 Thread nakaji
Hi, I have a question about jitterbuffer and PLC. I use Asterisk 1.6.2.0 and 1.6.0.20 or older. I use uLaw. My system map: = [ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ]

[asterisk-users] Question about PLC of Asterisk

2010-01-06 Thread nakaji
Hi,I want to know how to do to work PLC of Asterisk. Anyone plz help me. PLC (Packet Loss Concealment) is included in Asterisk,I read at voip-info.org or release note. And I see in codecs.conf, genelicplc setting. So I put codecs.conf in '/etc/asterisk' ,and wrote genericplc = true. And I