Displaying 9 results from an estimated 9 matches for "actarg".
2007 Mar 26
2
Polycom 601 loop
...26 09:51:20 DEBUG[6208] pbx.c: Function result is '(null)'
Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '1174924280.41882'
Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '(null)'
For a complete log (1.7 mb) of a single call to the extension, see
http://www.actarg.com/all_log
The polycoms are running bootrom 3.2.2.0019 and application version
1.6.7.0098. Any help on this would be greatly appreciated.
2007 Mar 26
1
SIP registration
...;sip:201@192.168.2.13>' failed for '192.168.3.2' - Not a local SIP domain
In sip.conf I have this for my global settings:
[general]
context=from-sip ; Default context for incoming calls
; if asterisk was compiled with OSP support.
realm=actarg.com ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain name
bindport=5060...
2007 Mar 30
0
Redirect failed, channel not up.
When I use the Asterisk manager interface to redirect a call (Action:
Redirect) I get an error with the message "Redirect failed, channel not up."
This is especially troubling as it looks like this message was added to
the code for the rather recent 1.2.x release. A quick google search
implies that I'm not the only one experiencing this problem with 1.2.17,
but me and
2006 Jun 06
1
Asterisk exit on startup
I'm having a problem with a new installation of asterisk 1.2.5 with a
digium dual port T1 (span 1 connected to an outside line, and span 2
connected to a CAC access bank I channel bank with 24 fxs ports). When I
start Asterisk (either from safe_asterisk or asterisk -vvvc) it will
immediately exit after it initializes. It will start the logger,
register applications and functions, register
2007 Mar 08
3
Sender phone ringing while recipient talking
I've had asterisk running for about a month now between our PBX and our
T1, and everything seems fine but for one simple nit-pick: When a call
to the outside workd is made, and if the recipient picks up while a the
sender's phone is still relaying the ring, the sender won't be heard
until after the ring stops. This often translates a simple "hello?" into
a
2007 Mar 30
1
call file vs. originate
I'm having trouble getting the manager interface to behave properly;
specifically the Originate event.
If I create an originate event as below, the calling phone will
auto-answer (as it's supposed to) but the receiving phone never rings.
It will timeout at 20 seconds.
Action: Originate
Channel: Local/201@from-sip2
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe
2006 Jun 09
2
T1 passthrough/middleman
Is it possible to act as a middle man on a T1 line?
My installation currently has an aging Inter-Tel Axxess box with a T1
coming in (16 in, 8 out). Rather than adding and replacing phones and
cards as they die, I would like to slowly migrate to a asterisk SIP
installation.
I want to take the incoming T1 line, use any available outgoing lines
for outgoing SIP, intercept any incoming lines and
2006 Nov 12
1
outgoing works, incoming fails on asterisk passthrough to inter-tel
Hi everybody,
Well, I've finally got asterisk to to talk nicely with my Intertel pbx.
Currently there is a outside T1 line (e&m wink start, esf, b8zs)
connected to asterisk, and then asterisk connected similarly to my
Intertel pbx. For right now all asterisk is doing is passing calls
between the two.
When I call out from the pbx, I can connect perfectly to the outside
world. When I