[Serusers] SER and Asterisk

[Rjey Nomer]

Hi all,

Hope someone can help me with this.

The main objective are to make Asterisk and SER
communicate with each other. Call SER--> Asterisk and
Asterisk--> SER.

I used the example configuration for pstn as base on
all the docs, it is how we can make ser and asterisk
work. As instructed on the docs, we just need to add
the IP address of the PSTN gateway on the trusted
database of our SER, in this case, asterisk is our
PSTN gateway. 

===================
mysql> insert into trusted values
("192.168.1.247","any","^sip:.*$");  
===================

On the asterisk (trixbox) server, below are the
configuration I defined:

Outgoing Setting:
Trunk Name: serout
Peer Details:
====================
allow=all
dtmfmode=rfc2833
host=192.168.1.41
insecure=no
type=peer
====================

I can ring any number on my SER Server using any 
number on my Asterisk, problem is when I pick-up the
phone I cannot hear voice and according to the logs as
listed below, it is hanging-up and something like
"Unresolvable destination".


SER NGREP RESULT
#ngrep -n 5060 -d eth0 3242194ngrep -n 5060 -d eth0
3242194
=================================
U 192.168.1.41:5060 -> 192.168.1.247:5060
SIP/2.0 478 Unresolvable destination (478/TM)..Via:
SIP/2.0/UDP
192.168.1.247:5060;branch=z9hG4bK580c7aec;rport=5060..F
  rom: "3000"
<sip:3000 at 192.168.1.247>;tag=as7693144e..To:
<sip:3242194 at 192.168.1.41>;tag=419e8..Call-ID:
5784a2681edd513
  c59c77e20512b499d at 192.168.1.247..CSeq: 103
INVITE..Server: Sip EXpress router (0.9.3
(i386/linux))..Content-Length: 0..
  Warning: 392 192.168.1.41:5060 "Noisy feedback
tells:  pid=9413 req_src_ip=192.168.1.247
req_src_port=5060 in_uri=sip:3
  242194 at 192.168.1.6:52961;user=phone
out_uri=sip:3242194 at 192.168.1.6:52961;user=phone
via_cnt==1"....
=================================

Asterisk Logs
=================================
  -- Executing NoOp("SIP/3000-08fd48b8", "CallerID set
to "3000" <3000>") in new stack
    -- Executing Set("SIP/3000-08fd48b8",
"GROUP()=OUT_2") in new stack
    -- Executing GotoIf("SIP/3000-08fd48b8", "0?108")
in new stack
    -- Executing Set("SIP/3000-08fd48b8",
"DIAL_NUMBER=3242194") in new stack
    -- Executing Set("SIP/3000-08fd48b8",
"DIAL_TRUNK=2") in new stack
    -- Executing AGI("SIP/3000-08fd48b8",
"fixlocalprefix") in new stack
    -- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix
  fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf
    -- AGI Script fixlocalprefix completed, returning
0
    -- Executing Set("SIP/3000-08fd48b8",
"OUTNUM=3242194") in new stack
    -- Executing Set("SIP/3000-08fd48b8",
"custom=SIP/Serout") in new stack
    -- Executing GotoIf("SIP/3000-08fd48b8", "0?16")
in new stack
    -- Executing Dial("SIP/3000-08fd48b8",
"SIP/Serout/3242194|120|r") in new stack
    -- Called Serout/3242194
    -- SIP/Serout-08fda208 is ringing
    -- SIP/Serout-08fda208 answered SIP/3000-08fd48b8
    -- Attempting native bridge of SIP/3000-08fd48b8
and SIP/Serout-08fda208
    -- Got SIP response 478 "Unresolvable destination
(478/TM)" back from 192.168.1.41
  == Spawn extension (macro-dialout-trunk, s, 14)
exited non-zero on 'SIP/3000-08fd48b8' in macro
'dialout-trunk'
  == Spawn extension (macro-dialout-trunk, s, 14)
exited non-zero on 'SIP/3000-08fd48b8'

=================================

Can anyone help me resolve this problem.

Thanks in advanced.

Rjey


       
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