Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?

2008-01-20 Thread William Stillwell (Ki4swy)
I have been running 1.4.17 since its release, and no kernal panics. 

Before that I was running 1.4.13 without any kernal panics. 

System Specs: 
4 Core Xeon 5110 @ 1.6Ghz (two dual proc chips) 
8 Gb Ram 
400GB Raid 5 SAS Array 

-- Original Message --
From: Ira [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Date:  Fri, 18 Jan 2008 12:20:56 -0800

At 11:53 AM 1/18/2008, you wrote:

Apart from the fact asterisk 1.2 is in security maintenance
mode only and wont get any other bugfixes it will be ok.
Please consider using 1.4 as it's the official latest stable
version.

Although for some of us, or at least me, no version of 1.4 has run 
for more than 72 hours before generating a kernel panic. I've tried 
about 6 versions, the early ones were good for about 10 minutes, the 
latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2.

Ira 


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[asterisk-users] Samsung iDCS 500R2 PRI Asterisk 1.4.*

2007-12-27 Thread William Stillwell (Ki4swy)
I'm having trouble getting an Asterisk server to make outbound calls on my iDCS 
Trunks.

I have idcs station to asterisk station working
I have asterisk station to idcs station working

However, I am unable to get Asterisk to utilize any outbound trunks on my 
iDCS

Anybody have any ideas?

 





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[asterisk-users] Recommend Digium Hardware?

2007-09-28 Thread William Stillwell (Ki4swy)
What is the recommend Digium Card for a PRI in NA ?

I want to interface a Asterisk Server to a Samsung iDCS System, and have 
available T1 w/DNIS, or a PRI w/DID, the asterisk server would need to appear 
as a Telco provided Circuit.

Slot Availability.
Four PCI-Express Slots x8 (1 full-length/1 half-length/2 low-profile).

 





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[asterisk-users] Mutipoint Conferencing?

2007-09-14 Thread William Stillwell (Ki4swy)
I am trying to determine what would need to be done/modified to enable the 
following:

I have a SIP extension come into my asterisk box, and I then need it to call 
6-10 remote Sip Stations that are set to Auto-Answer...

(note, my remote sip stations are actually cisco h323 devices, I can call them 
fine from any softphone, or other device, and have full-duplex audio, however, 
i need to be able to conference bring all the remote stations 
automatically.w/Full duplex audio.

Or if someone could direct me to a list that would actually be able to answer 
this question..

Thanks,

W. Stillwell 





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[asterisk-users] Grandstream GXW-4104 ???

2007-09-02 Thread William Stillwell (Ki4swy)
How well does the Grandstream GXW-4104 or (8) work w/Asterisk? I would use a 
Cisco Switch w/FXO Ports but that would be a little Pricy

I Can't use a Digium FXO Card, as the asterisk Server is offsite.

Thanks,
William Stillwell
KI4SWY
 





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[asterisk-users] Wireless VOIP Keysets? Recommendations?

2007-09-02 Thread William Stillwell (Ki4swy)
Any Recommendations on a Good Wireless Voip Keyset that works well with 
Asterisk?

I would prefer one that is IAX2 as it works better behind a Nat'd Firewall.. 

But I am reaching out to you guys as you all would know what would work the 
best :-)
 





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[asterisk-users] Cisco FXS Issue...

2007-08-29 Thread William Stillwell (Ki4swy)
Im sure this has been thrown around this list 1,000 times, and Im sure its been 
around the net too.. But I have done everything, and cannot seem to get inward 
calls to be processed on my asterisk box..

First, Let me tell you what works:

1) Softphones (ZoIPer using IAX2 Protocol) Can make calls behind a Natted 
Firewall to the FXS Port, and it rings, and calls work full duplex.

2) Soyo IP Phone (Loaded with IAX2 Firmware) Can make calls on the same subnet 
as the asterisk server, and can talk full duplex to the FXS Port.

3) I can call other FXS Ports on the same Switch.

 Hardware 

Cisco 3725 w/ Two FXO and Two FXS

Config:

voice-port 2/1/0
 mwi
 description FXS Port 0
 station-id name fxs_2_1_0
 station-id number 1000
!
voice-port 2/1/1
 mwi
 description FXS Port 1
 station-id name fxs_2_1_1
 station-id number 1001
!
!
!
!
!
dial-peer voice 1000 pots
 description Binds to FXS Port 2/1/0
 destination-pattern 1000
 port 2/1/0
 authentication username 1000 password password
!
dial-peer voice 1001 pots
 description Binds to FXS Port 2/1/1
 destination-pattern 1001
 port 2/1/1
 authentication username 1001 password password
!
dial-peer voice 200 voip
 destination-pattern .T
 progress_ind progress enable 8
 session protocol sipv2
 session target ipv4:ip of asterisk server:5060
 session transport udp
 codec g711ulaw

sip-ua 
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 registrar ipv4:ip of asterisk server expires 60
 sip-server ipv4:ip of asterisk server:5060
 notify telephone-event max-duration 500



 Asterisk Info -

sip.conf

[1000]
  username=1000
  type=friend
  secret=password
  qualify=yes
  nat=no
  insecure=very
  Host=dynamic
  dTMFMode=rfc2833
  auth=md5,plaintext
  allow=ulaw


[1001]
  username=1001
  type=friend
  secret=password
  qualify=yes
  nat=no
  insecure=very
  Host=dynamic
  auth=md5,plaintext
  allow=ulaw


--- 

The cisco box does register.

Dialing anything on the fxs port results in Fast Busy with no warnings/errors 
showing up console. 





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