[Serusers] SER and Asterisk

Andrey Kuprianov andrey.kouprianov at gmail.com
Tue Jun 5 13:54:07 CEST 2007 

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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