[Serusers] SER and Asterisk

Andrey Kuprianov andrey.kouprianov at gmail.com
Tue Jun 5 13:54:40 CEST 2007 

In no.3 i meant Trixbox, not Asterisk :)

On 6/5/07, Andrey Kuprianov <andrey.kouprianov at 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 at 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 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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> >
> >
> >
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