Yo!
I changed callprogress to no, and in wcfxo.c source around line 334 i changed 
the value 32000 and -32000 to 10000 and -10000 because it had something to do 
with the DC voltage when it was ringing.
I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an 
interesting diagram of wiring that was incorrect for sending voltage to a 
phone or something like that.

So put it in your asterisk knowledge bases.
Cheers, have a good weekend
----------  Forwarded Message  ----------

Subject: SIP phones can't pick up incoming call on analog trunk - signalling 
problem?
Date: Friday 13 January 2006 13:50
From: C Mylo <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com

A very good day to you all,

We can't get the phones to pick up on an incoming call on analog trunks.
We're using the digium products in the box, with snom phones internally.

This is the output from the asterisk console:
linux*CLI> zap show channels
   Chan Extension  Context         Language   MusicOnHold
 pseudo pstn-incoming   en         default
      1            pstn-incoming   en         default
      2            pstn-incoming   en         default
      3            pstn-incoming   en         default
      4            pstn-incoming   en         default
    -- Starting simple switch on 'Zap/1-1'
Jan 13 13:32:13 WARNING[25905]: chan_zap.c:6105 ss_thread: CallerID returned
with error on channel 'Zap/1-1'
    -- Executing Dial("Zap/1-1", "SIP/24|15") in new stack
    -- Called 24
    -- SIP/24-406f is ringing
    -- SIP/24-406f is ringing
    -- SIP/24-406f is ringing
    -- Starting simple switch on 'Zap/4-1'
Jan 13 13:32:16 NOTICE[25908]: chan_zap.c:6031 ss_thread: Got event 18 (Ring
Begin)...
    -- SIP/24-406f is ringing
Jan 13 13:32:18 NOTICE[25908]: chan_zap.c:6031 ss_thread: Got event 2
(Ring/Answered)...
    -- Executing Dial("Zap/4-1", "SIP/24|15") in new stack
    -- Called 24
    -- SIP/24-c8de is ringing
    -- SIP/24-c8de is ringing
  == Spawn extension (pstn-incoming, s, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
    -- SIP/24-c8de is ringing
Jan 13 13:32:21 WARNING[25908]: chan_zap.c:3904 zt_handle_event:
 Ring/Off-hook in strange state 6 on channel 4
    -- SIP/24-c8de is ringing
Jan 13 13:32:24 WARNING[25908]: chan_zap.c:3904 zt_handle_event:
 Ring/Off-hook in strange state 6 on channel 4
    -- SIP/24-c8de is ringing
  == Spawn extension (pstn-incoming, s, 1) exited non-zero on 'Zap/4-1'
    -- Hungup 'Zap/4-1'
    -- Starting simple switch on 'Zap/4-1'
Jan 13 13:32:31 NOTICE[25916]: chan_zap.c:6031 ss_thread: Got event 18 (Ring
Begin)...
Jan 13 13:32:33 NOTICE[25916]: chan_zap.c:6031 ss_thread: Got event 2
(Ring/Answered)...
    -- Executing Dial("Zap/4-1", "SIP/24|15") in new stack
    -- Called 24
    -- SIP/24-9f09 is ringing
    -- SIP/24-9f09 is ringing
    -- SIP/24-9f09 is ringing
    -- SIP/23-b60e answered Zap/4-1
Jan 13 13:32:36 WARNING[25916]: chan_zap.c:3904 zt_handle_event:
 Ring/Off-hook in strange state 6 on channel 4
  == Spawn extension (pstn-incoming, s, 1) exited non-zero on 'Zap/4-1'
    -- Hungup 'Zap/4-1'


& zapata.conf

[channels]
language=en
toneduration=100
context=pstn-incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
usecallingpres=yes
threewaycalling=yes
transfer=yes
canpark=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
flash=100
transfertobusy=yes
busydetect=yes
busycount=4
busypattern=400,400
callprogress=yes
progzone=au
ringtimeout=6000
faxdetect=incoming
musiconhold=default
channel=>1 ;X100P
channel=>2 ;X100P
channel=>3 ;X100P
channel=>4 ;X100P

& zaptel.conf

fxsks=1-4
loadzone = au
defaultzone=au

& extensions.conf
...relevant part...
[pstn-incoming]
exten => s,1,Dial(SIP/24|15)
exten => s,n,Dial(SIP/21&SIP/22&SIP/23&SIP/24&SIP/25&SIP/26&SIP/27|120)


My experience with asterisk isn't with analog signalling and it has been a
long time since I looked at zapata.conf with a keen eye.  I've never not been
able to pick up a call so I'm a little baffled and that's why i'm presuming
it's something to do with signalling.

Thanks

-------------------------------------------------------

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to