There is error correction in TFTP. Its done at the application layer and not
the transport layer.
TFTP uses two UDP ports for control and data transfer, this is probably
where there are problems with NAT devices.
The control connection is ;
client - sport dynamic(x) -> server dport 69
client
Hi
I've got a similar problem.
Asterisk 1.0.7 running on Dual Opeteron smp Suse linux.
Zaptel taken from CVS yesterday with USE_RTC enabled (no Zaptel hardware)
With three users on a conference I see about a 1/2 second delay, it like
having a call across a satellite link.
'iax2 show peers' s
make sure you have 'canreinvite=no' in sip.conf
Cheers
Giles
- Original Message -
From: "Derek Conniffe" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Monday, May 23, 2005 5:39 PM
Subject: [Asterisk-Users] ZyXEL Prestige 2000W - cant make
Hi
Has anyone got a Spectralink SVP server to talk to
Asterisk?
Aparently they talk h.323 with g.711
codec.
Cheers
Giles
--
This message has been scanned for viruses and
dangerous content by
www.swiftinter.net, and is
believed to be clean.
___
Nortel AAS-2000 range of LB's can do this today.
Giles
- Original Message -
From: "Patrick" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, February 03, 2005 12:38 PM
Subject: Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
Hi,
Does anyone have a GS101 version of the ringtone
used in the Fox show 24?
I think the ringtone is from a cisco
phone?
Cheers
Giles--
This message has been scanned for viruses and
dangerous content by
www.swiftinter.net, and is
believed to be clean.
___
Hi
I did some testing with AirCrack against a Senao si-7800 and an AP (WEP
128bit key).
Aircrack cracked the 128bit WEP key after 20 minutes. (there was a
continuous voice call going on during that period).
Senao Wifi phones leak weak IV's once a minute or so (Airsnort would take
days to crack
Hi
Just received a couple of SI-7800 wifi
phones.
nice looking phone, got it to work after a bit of a
headache, which I thought I would share.
sip.conf
[1007]type=friendusername=1007secret=blahhost=dynamiccontext=from-sipdisallow=allallow=ulaw
The phone has a problem selecting codec's
heers
Giles Scott
Hi,
With my config (as posted this morning) DTMF works.
I can log onto voicemail by selecting a mailbox number and password
Giles
- Original Message -
From: "Dominique Kull" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, June 24, 2004 12:02 PM
Subject: [Asterisk-Users] ZyXE
ure why, makes the
> same tritone as the mobile when it's impossible to place a call).
> Would you mind if I put your recipe on voip-info?
> Thanks,
> l.
>
>
>
> In data Thu, 24 Jun 2004 10:34:48 +0100, Giles Scott
> <[EMAIL PROTECTED]> ha scritto:
>
> >
Hi
I got it to work straight off (i've since upgraded the code to WJ.00.0b-t04)
sip.conf
[1003]
type= friend
username=1003
secret=blah
host=dynamic
context=from-sip
dtmfmode=rfc2833
Zyxel config
SIP/outbound Proxy config
Proxy IP:192.168.254.1
Proxy port = 5060
SIP Config
SIP UII sip: 1003 @ 19
Hi,
Has anyone written a management application
(WinXP) which can display simple Queue Stats on a
monitor.
I would like to see;
No of Calls currently in the queue
Hold time
Completed
Abandoned
I know the info is available via API,'Action:
QueueStatus', I just don't want to re-invent th
13 release date Apr 12 2004
Cheers
Giles Scott
- Original Message -
From: "Dominique Kull" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, June 02, 2004 3:46 PM
Subject: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
> Does anybody have any
14 matches
Mail list logo
