Hi, Richard,
I still try, but fail with rtp transfer.
2005/9/27, richard Coco <[EMAIL PROTECTED]>:
>
> > I still find out how to let LCS 2005 accept SIP
> > invite from Asterisk,
> > Need more help.
>
> Hi jacky,
>
> can you please share your experience
Search google with "sip pstn site:www.microsoft.com"
You will find out how to configure LCS static routing to SIP Gateway,
like Asterisk
but you need patch Asterisk to support TCP.
http://bugs.digium.com/view.php?id=4903
Step1: configure LCS 2005 to let sip uri: [EMAIL PROTECTED] to route to
next
LCS 2005 just support SIP TCP or TLS right now.
so you must patch asterisk chan_sip.c support TCP,
look http://bugs.digium.com/view.php?id=4903
I have successful call to asterisk's SIP peer or PSTN use Office
Communicator 2005(sign-in my LCS 2005)
but I can't use Dial(SIP/[EMAIL PROTECTED]) , let
;
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
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he problem when softswitch G.729A and G.729B codec.
What source code I must to modify ?
Or some people have finished the issue, Could you show me how to do?
--
Jacky
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Hi, List
The Gentoo portage tree only include 0.9.0, it seems no upgrade for long time.
Do you know someone have the 1.0.1 ebuild version?
--
Jacky
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audio codec?
Does Asterisk only bypass the codec frame when call is not softswitch?
Can * handle mpeg4 or other codec when video client use this codec?
Thanks,
--
Jacky
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Hi All,
I which to use Daemontools to watch asterisk process.
Step.1
After install daemontools, i add a line in /etc/inittab
SV:123456:respawn:/usr/local/bin/svscanboot
Step.2
and create /etc/asterisk/run
#!/bin/bash
echo -n "Starting Asterisk PBX: "
exec /usr/sbin/asterisk
Step.3
ln -sf /et
Hello, Hacker
I install VoiceTronix OpenSwitch 12 port PCI Telephone Card,
and setting vpb.conf, extensions.conf following
My problem is:
When i dial to fxo(channel 9-12), it is ok,
but when i continue press exten 102, the channel crach with error messages
following
exception caught: VPBAPI_DIAL
MessageHi, All,
I have buy a Asterisk Developer's Kit, there include Wildcard X100P &
TDM400P(10B)
The TDM400P module is RJ45 connector(8 line Jack), but RJ11 analog phone
only need 2 line
Is it possible to connect TDM400P module1 with four RJ11 analog phone(use
cat.5 network line)
ex. pin1,2 ->
Hi, all
I have builded a pbx server for pstn, sip & h.323 users
but i can't find any example extensions.conf for access
control when users which call longdistance with pstn,
If anyone have good example, please sharing your experience
Thanks very much
___
The extension.conf:
exten => s,1,Wait,1 ; Wait a second, just
for funexten => s,2,Answer ; Answer the lineexten
=> s,3,DigitTimeout,5 ; Set Digit Timeout to 5 secondsexten
=> s,4,ResponseTimeout,10 ; Set Response Timeout to 10
secondsexten => s,5,BackGround,demo-congrats ; P
hanks!
jacky qiao
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