5 Jun
2007
5 Jun
'07
2:54 p.m.
In no.3 i meant Trixbox, not Asterisk :)
On 6/5/07, Andrey Kuprianov <andrey.kouprianov(a)gmail.com> wrote:
Can you try these things for Asterisk:
1. nat=yes
2. insecure=very (just try and see if this works)
3. Since you are using Asterisk, try to set Allow anonymous SIP calls
to "yes" in the general settings of FreePBX interface
On 6/5/07, Rjey Nomer <rjeynomer(a)yahoo.com> wrote:
I already define the hostname of both machines
and I
created also an internal DNS wherein I define the
following:
==================================
# dig -t SRV _sip._udp.rjey.ph
; <<>> DiG 9.2.4 <<>> -t SRV _sip._udp.rjey.ph
;; global options: printcmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id:
13180
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 3, AUTHORITY:
2, ADDITIONAL: 5
;; QUESTION SECTION:
;_sip._udp.rjey.ph. IN SRV
;; ANSWER SECTION:
_sip._udp.rjey.ph. 10800 IN SRV 0 0 5060
asterisk.rjey.ph.
_sip._udp.rjey.ph. 10800 IN SRV 0 0 5060
ser.rjey.ph.
;; AUTHORITY SECTION:
rjey.ph. 10800 IN NS ns1.rjey.ph.
rjey.ph. 10800 IN NS ns2.rjey.ph.
;; ADDITIONAL SECTION:
ser.rjey.ph. 10800 IN A 192.168.1.41
asterisk.rjey.ph. 10800 IN A 192.168.1.247
ns1.rjey.ph. 10800 IN A 192.168.1.7
;; Query time: 46 msec
;; SERVER: 192.168.1.7#53(192.168.1.7)
;; WHEN: Tue Jun 5 18:32:58 2007
;; MSG SIZE rcvd: 292
============================
What else should I do????
Thanks,
Rjey
Date:
Tue, 05 Jun 2007 07:24:27 -0400
From:
"SIP" <sip(a)arcdiv.com> Add to Address Book Add
Mobile Alert
To:
"Rjey Nomer" <rjeynomer(a)yahoo.com>
CC:
serusers(a)iptel.org
Subject:
Re: [Serusers] SER and Asterisk
This means you can't resolve 192.168.1.247 from your
SER server. Try
adding an entry for it into /etc/hosts to see if you
can bypass DNS.
N.
Rjey Nomer wrote:
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@192.168.1.247>;tag=as7693144e..To:
<sip:3242194@192.168.1.41>;tag=419e8..Call-ID:
5784a2681edd513
c59c77e20512b499d@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@192.168.1.6:52961;user=phone
out_uri=sip:3242194@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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