[Serusers] SER and Asterisk
Rjey Nomer
rjeynomer at yahoo.com
Tue Jun 5 13:47:05 CEST 2007
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 at arcdiv.com> Add to Address Book Add
Mobile Alert
To:
"Rjey Nomer" <rjeynomer at yahoo.com>
CC:
serusers at 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 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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