Re: [Asterisk-Users] Satellite Broadband and VOIP

2005-09-26 Thread Mark Anthony C. Delfin

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
 




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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

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[Asterisk-Users] [EMAIL PROTECTED] newbie extensions always busy

2005-08-02 Thread Mark Anthony C. Delfin

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

--
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Mark Anthony C. Delfin


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Re: [Asterisk-Users] [EMAIL PROTECTED] newbie extensions always busy

2005-08-02 Thread Mark Anthony C. Delfin
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


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--
__
Mark Anthony C. Delfin
Satellite Engineer
Textron Corporation
email: mcdelfin at itextron dot com
Tel + (632) 726 6164
Fax + (632) 724 1916 


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[Asterisk-Users] newbie, error 401 unauthorzed question

2004-06-18 Thread Mark Anthony C. Delfin
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=welltech801
   
   
[x802]
  type=friend
  username=802
  secret=12345
  host=dynamic
  dtmfmode=inband
  canreinvite=no
  callerid=antek802

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:  sip:[EMAIL PROTECTED]:5060
To: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 15 REGISTER
Contact: sip:[EMAIL PROTECTED]:5060
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: sip:[EMAIL PROTECTED]:5060
To: sip:[EMAIL PROTECTED]:5060;tag=as6e12e2cb
Call-ID: [EMAIL PROTECTED]
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
 
 
 to 210.16.20.14:5060
Jun 19 12:43:41 NOTICE[-1116562512]: chan_sip.c:5623 handle_request: Registration from 
'sip:[EMAIL PROTECTED]:5060' failed for '210.16.20.14'

Thanks in Advance

Mark


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