Edit:
I went ahead and created a thread under support for this: viewtopic.php?f=1&t=72205
Can some moderator please remove this thread?
Thanks!
Hello all,
I am trying to get a set of three Polycom ip 321’s working. I got a Digium 410 card with two FXO ports. I am able to call out and receive calls. I have a Debian stable system with asterisk 1.4.21.2~dfsg-3 and zaptel 1.4.11~dfsg-3.
Here are the problems I am having:
-
The phone does not show when the line is busy. Is there any way to do this?
-
The phone rings whenever anything is done on the analog line. (The phones lines come in and are split into both asterisk and the analog system). This is not gonna be a huge problem once most of the phones are replaced with ip phones however we are still gonna have our fax machine. The phone will ring if someone on the analog side picks up a phone to dial and whenever they finish a call.
So far I just have one phone out of the box and have given it two lines. Is there any better way to do this then registering twice? My sip.conf
[code][sec]
type=friend
context=line_one_internal
host=dynamic
secret=foo
[sec2]
type=friend
context=line_two_internal
host=dynamic
secret=foo[/code]
My zaptel config file looks like:
[code]# Autogenerated and some other comments
fxsks=1
fxsks=2
channel 3, WCTDM/0/2, no module.
channel 4, WCTDM/0/3, no module.
Global data
loadzone = us
defaultzone = us
[/code]
my zapata is:
[code][trunkgroups]
; define any trunk groups
[channels]
; hardware channels
; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
; define channels
signalling=fxs_ks
immediate=yes
context=line_one
channel => 1
context=line_two
channel => 2[/code]
Anyone know whats I should do?
Thanks!
Edit: I can’t seem to add extension.conf for you to look at? If you want more information on whats in it let me know.
Edit2: Aww great I got this stuck in general instead of support. Any way to move it?