I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and
a couple computers with eyebeam. I have one small. I cannot call the
eyebeam clients from the phone connected the fxs port. I can call the
phone from the eyebeem clients. And, I get both the fxs phone and
eyebeam clients to ring when a call comes in through the fxo port.
I have been trying to get this straightened out for quite a while and
have tried suggestions in the wikis and mailing lists but haven't had
any luck so far.
The output from the console is:
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", ""call for: " 3000") in new stack
-- Executing Dial("Zap/1-1", "SIP/3000|60|tr") in new stack
-- Called 3000
Aug 25 10:16:13 NOTICE[4092]: chan_sip.c:1806 auto_congest:
Auto-congesting SIP/3000-d838
-- SIP/3000-d838 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing VoiceMail("Zap/1-1", "u3000") in new stack
-- Playing '/var/spool/asterisk/voicemail/default/3000/unavail'
(language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (from-pots-internal, 3000, 3) exited non-zero on
'Zap/1-1'
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (from-pots-internal, h, 1) exited non-zero on
'Zap/1-1'
-- Hungup 'Zap/1-1'
Aug 25 10:16:21 WARNING[4092]: chan_sip.c:1055 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
Aug 25 10:16:25 WARNING[4092]: chan_sip.c:1055 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Request)
zapata.conf
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
rxgain=0.0
txgain=0.0
;callgroup=1
;pickupgroup=1
immediate=no
busydetect=yes
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number
is in milliseconds
callerid=asreceived
signalling=fxs_ks
group=2
context=from-analog ; Points to the incoming context of your
extensions.conf
channel => 4
signalling=fxo_ks
callerid="Paul Wolstenholme" 604.267.2556
group=1
context=from-pots-internal
channel=>1
sip.conf
[general]
port=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0 ; Address to bind to (all addresses on
machine)
context=from-sip-external ; Send unknown SIP callers to this context
[3000]
type=friend
username=3000
secret=9876
host=dynamic
defaultip=192.168.1.100
context=from-sip-internal
mailbox=3000
nat=no
invite=no
canreinvite=no ; Leave this alone for now; see archives for
details
qualify=1000
;dtmfmode=inband
dtfmode=rfc2833 ; inband is not supported in compressed codecs like
gsm, so we better set it to rfc2833
disallow=all
allow=gsm
extensions.conf
[local-sip-extensions]
exten => 3000,1,NoOp("call for: " ${EXTEN})
exten => 3000,2,Dial(SIP/3000|60,tr)
exten => 3000,3,Voicemail(u3000)
exten => 3000,102,Voicemail(b3000)
exten => 3000,103,Hangup
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