Re: [Asterisk-Users] Satellite Broadband and VOIP
Hi Sean, We operate a VSAT network here in the Philippines (using Shiron, FDMA Bandwidth on Demand) and offer VoIP using asterisk. We do not sell our voip to our gilat clients since gilat has a higher latency (since it uses TDMA). Try to look for a satellite provider that has an average (to your country of voip destination) latency of below 600-800 ms and it must be consistent. Also since, satellite has low upload bandwidth, try to have QoS behind the satellite modem and prioritize VoIP traffic cxpcman wrote: Sean Rima wrote: I live in a very rural area, BB access will never happen and the only choice I have it Satellite. I seen from a post to this list that Gilat sat modems are not recommended. Is this still the case or is there another alternative? Sean ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well is not recommended because of the seektime . the information you send and recive have a delay no matter how fast your conection is .. so you gonna hear the voice out of time . wire have a lot faster response times than air soo... ur choice ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] newbie extensions always busy
Thanks for the info, now i'm using extensions 200 and 201 and disabled voicemail but when I try to calling one another it returns busy Howard Leadmon wrote: Well not sure if this would cause the problem, but do know that by default AAH uses the 8XXX series numbers for the conference bridge. So I would think putting extensions in the 8 range would be a bad idea, but someone on here may tell me this is wrong. --- Howard Leadmon - [EMAIL PROTECTED] http://www.leadmon.net -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Anthony C. Delfin Sent: Tuesday, August 02, 2005 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] newbie extensions always busy hi list, I'm running a newly installed [EMAIL PROTECTED] an i registered two soft phone. both soft phone are registered 8901/8901x.x.x.xD 255.255.255.255 50710Unmonitored 8900/8900y.y.y.y D 255.255.255.255 6281 Unmonitored but when I call one another, they are always busy and directed to its voicemail Sorry, if this was posted before TIA -- __ Mark Anthony C. Delfin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __ Mark Anthony C. Delfin Satellite Engineer Textron Corporation email: mcdelfin at itextron dot com Tel + (632) 726 6164 Fax + (632) 724 1916 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] newbie extensions always busy
hi list, I'm running a newly installed [EMAIL PROTECTED] an i registered two soft phone. both soft phone are registered 8901/8901x.x.x.xD 255.255.255.255 50710Unmonitored 8900/8900y.y.y.y D 255.255.255.255 6281 Unmonitored but when I call one another, they are always busy and directed to its voicemail Sorry, if this was posted before TIA -- __ Mark Anthony C. Delfin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie, error 401 unauthorzed question
hello asterisk list, i've installed asterisk-0.9.0 on fedora core 1, i've been receiving error 401 when i connect my voip gateways on asterisk (welltech fxo, antek fxs) here is my sip.conf [general] port=5060 bindaddr=0.0.0.0 context=from-sip tos=lowdelay [x801] type=friend username=801 secret=12345 host=dynamic dtmfmode=inband canreinvite=no callerid="welltech"<801> [x802] type=friend username=802 secret=12345 host=dynamic dtmfmode=inband canreinvite=no callerid="antek"<802> here is my extensions.conf [general] static=yes writeprotect=no [globals] welltechfxo=SIP/x801 antekfxs=SIP/x802 [from-sip] include => to-sip [to-sip] exten => 801,1,Dial(${welltechfxo},20,tr) exten => 802,1,Dial(${antekfxs},20,tr) here is the debug output Sip read: REGISTER sip:210.16.20.7:5060 SIP/2.0 Via: SIP/2.0/UDP 210.16.20.14:5060;branch=0 From: To: Call-ID: [EMAIL PROTECTED] CSeq: 15 REGISTER Contact: Expires: 60 Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 210.16.20.14 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 210.16.20.14:5060;branch=0 From: To: ;tag=as6e12e2cb Call-ID: [EMAIL PROTECTED] CSeq: 15 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 210.16.20.14:5060 Jun 19 12:43:41 NOTICE[-1116562512]: chan_sip.c:5623 handle_request: Registration from '' failed for '210.16.20.14' Thanks in Advance Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users