Another IP phone possibility for Asterisk.
No, not the SPA941 (from the Linksys/Cisco/Sipura world)...
Don't know much about it... but found this. Nothing on the datasheet
says what it'll support really.
http://newsroom.cisco.com/dlls/2005/eKits/Data_Sheet_IP_Manager_Phone.pdf
But I found
VOIP pioneer predicts a roiling 2005 for IP telephony
Eetasia.com (subscription) - USA
Open source software communications will begin to influence the VoIP market
in a big way next year, according to VoIP pioneer Jeff Pulver. ...
http://www.eetasia.com/article_content.php3?article_id=8800354924
I have my Linux 2.6 kernel with the necessary HDLC config and also recompiled
zaptel accordingly. On modprobe, I get:
Found a Wildcard: Digium Wildcard T100P T1/PRI
Debug: sleeping function called from invalid context at mm/slab.c:2000
in_atomic():0[expected: 0], irqs_disabled():1
[0211e605]
So Asterisk gurus out there, is there a nice clean way in the dialplan
to determine if the caller is coming from a transferred call, and on
the unavailable context in the dial, instead of going to e-mail go
back to the transferee?
If anyone has this sort of logic or could spit out an
These taken tonight (9/22/2004) at the Expo and Reception
Enjoy. http://photos.tropiano.org/gallery/astricon-2004
Lenny
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Does anyone know if the Marriott has Wi-Fi? LAN connection in the room?
According to the STSN (www.stsn.com) hotel locator, the Marriott does
have in room wired access. Wireless access and Meeting room access.
At $9.99/day (cheaper usually if you buy blocks of in multiple days)
locked to a
Did I see something on here about using an AGI script to do reverse
lookups via anywho.com? I have a PRI that only gets caller-id number and
no Alpha.
[...]
I put a copy of it here...
http://www.voiping.com/calleridnamelookup.agi
It was written by James Golovich [EMAIL PROTECTED]
We're doing some SIP development and have a question on additional parameters
supplied to the register (in this case maddr= and the non-standard clport= in
our example below).
What we're experiencing is the INVITE doesn't included these parameters
and they get dropped when the INVITE is sent to
I just read that Pingtel (www.pingtel.com) will be releasing it's IP PBX (which
runs under Linux) to open source (similar model to Redhat Linux, charging
for support, etc.). Read more about it at... http://www.pingtel.com/a_opensource.jsp
and http://www.tmcnet.com/usubmit/2004/Feb/1024036.htm
I really like the functionality that Gastman provides, it would solve a problem
I currently have that Secretaries don't know when/who's on the phone before they
transfer the caller...
But I'm seeing some oddities, maybe just because the code line hasn't
been updated in a while. Take a look
Looking for a current precompiled Win32 binary for gastman, don't
have a build environment for Windows. Also does gastman compile
under Linux and is there a current binary as well...
Thanks
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I've noticed I have an issue with my Dialplan ... apparently instead of a busy
signal when the caller is busy it falls through and gets a Congestion...
What's the proper syntax for this, reorder tone when there is a reorder and
busy when there is a busy...
SBC is a T1/PRI.
[macro-sbc-outdial]
Here's a simple small expect script ...
I call it phreboot, usage: phreboot IP
$ phreboot 10.99.1.1
-- cut here --
#!/usr/bin/expect -f
set timeout -1
spawn $env(SHELL)
match_max 1
send -- telnet [lrange $argv 0 0]\r
expect -exact word :
send -- cisco\r
expect -exact Phone
send --
I'm considering using the Agent login/logoff function to add to a queue
that will be our main number during the day to answer. Periodically
our receptionist is not at her desk and would be useful for her to
login elsewhere and get the main number calls to transfer as she sees
fit. If the
Folks --
I know this isn't directly an * issue, but I need to buy 14 7940s (preferably)
(or 7960s if the price is also reasonable) --- no power cubes, immediately. If
anyone has a good price, contact me offline at 512-427-1324 or
lenny @ rocksteady.com
Thanks,
Lenny
I'd like to play a sound to a user I dial (via SIP) once
they answer play the sound and then allow me to talk to them.
The new Cisco 7960 SIP code allows to set lines to autoanswer
via the speaker phone, I'd like to play a tone after it rings
through and then talk...
Any thoughts on how to do
gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586
-DASTERISK_VERSION=\CVS-03/25/03-10:49:30\ -DINSTALL_PREFIX=\\
-- Registered SIP '' at 192.70.239.2 port 5060 expires 3600
-- Executing Playback(SIP/lenny-4ee2, transfer|skip) in new stack
-- Executing Macro(SIP/lenny-4ee2, dial|7555|SIP/lenny-lap) in new stack
-- Executing Dial(SIP/lenny-4ee2, SIP/lenny-lap|20|tT) in new stack
-- Called
ZT_SIG_SF undeclared?
make[1]: Entering directory `/usr/local/src/asterisk/channels'
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586
-DASTERISK_VERSION=\CVS-03/12/03-21:24:47\
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