[Asterisk-Users] Internal FXS to SIP problem

[Paul Wolstenholme]

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 2347bee118aaed483f9d34a60a35b569 at 192.168.1.30 
for seqno 102 (Critical Request)
Aug 25 10:16:25 WARNING[4092]: chan_sip.c:1055 retrans_pkt: Maximum 
retries exceeded on call 2347bee118aaed483f9d34a60a35b569 at 192.168.1.30 
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