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Unread 03-30-2006, 02:04 PM   #1
v164
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Default Direct Dial IP

I have a SIP VoIP service bundled with my ISP in Japan. This VSP "blocks incoming SIP calls from non-business partner external domains" (as SIP Broker so consisely puts it). The only people who can call me through this VoIP service for free are other subscribers, in Japan, with the same ISP, or one of their business-partners.

(Edited 2007-03-03: Further, for someone to call me for free through this service, they need to know that I have this "IP Phone" service, and they need to know my 050xxxxxxxx phone number.)

My ATA however, will accept incoming calls directly from anywhere on the internet (any IP address). So if the person that wants to call me knows my IP address (and 050 number), they can call my ATA direct, for free (no need to traverse my VSP's proxy).

For example, using the X-Ten lite softphone, if I set "Direct Dial IP" to "Yes" (default is "No"), and then dial,

050xxxxxxxx@xxx.xxx.xxx.xxx

(xxx.xxx.xxx.xxx = my IP address)

the call goes straight to my ATA at home and my home phone rings.


I then went to www.e164.org, and set up an ENUM mapping. Both my home number (+8159xxxxxxx) and my VoIP number (+8150xxxxxxxx) are mapped to the SIP URI,

050xxxxxxxx@xxx.xxx.xxx.xxx

The test call at e164.org works fine, and using OzTell's "External free service" option in WebDialer also works (ie, using OzTell's webdialer, I dial my PSTN number, and the call comes arrives via VoIP). However, I haven't been able to get inbound calls to come in via SIP Broker, either using SIP Broker via VoIP, or using the SIP Broker PSTN gateway numbers.

So what I'm asking is, does SIP Broker / Voxalot work with Direct Dial IP?

Voxalot has a lot of configuration options - I was thinking maybe I could "call forward" my voxalot service to my IP address, or maybe I could setup my IP address as an external provider and transfer all calls there.

PS. It appears that my VSP blocks external SIP traffic at their routers. Hence, I can't register this VSP account to have it used with voxalot / OzTell webdialer etc.

Last edited by v164; 03-03-2007 at 01:27 AM. Reason: clarification
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Unread 03-30-2006, 08:59 PM   #2
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Hi v164,

Dracofelis has written a nice piece on doing exactly this on a Sipura / Linksys ATA. Hopefully there is something in there that might help you:

http://faq.sipbroker.com/tiki-index....%20or%20Sipura
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Unread 03-30-2006, 11:52 PM   #3
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Quote:
Originally Posted by martin
Dracofelis has written a nice piece on doing exactly this on a Sipura / Linksys ATA. Hopefully there is something in there that might help you:

http://faq.sipbroker.com/tiki-index....%20or%20Sipura
Agreed.

The OP didn't say what type of adapter was being used. However, the fact that the "test call" worked, means that the techniques used in my Wiki post should also work (but the details may be slightly different, depending upon what adapter the OP is used).

NOTE: I highly recommend the OP get a free SIP Broker alias (sign up here: http://www.sipbroker.com ), and point that "alias" at the dynamic DNS entry pointing to the OP's IP dialing address. Once that is done, it is trivial for the OP to be dialed via their SIP Broker alias (that now points directly to their adapter). For example, SIP Broker aliases can easily be dialed by: 1) Any Voxalot user (*011 alias), 2) any SIP Broker user (*011 alias), 3) any user of the free SIP Broker PSTN dialin numbers (*011 alias), 4) any user of a VoIP service that "peers" with SIP Broker (peer_code *011 alias), 5) and even anyone who can dial SIP URIs directly (*011alias@sipbroker.com).
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Unread 03-31-2006, 04:13 AM   #4
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Quote:
Originally Posted by DracoFelis
I highly recommend the OP get a free SIP Broker alias (sign up here: http://www.sipbroker.com ).
I don't see where you click to get the alias on the sipbroker.com page.
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Unread 03-31-2006, 06:06 AM   #5
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Hi,

Just register with sipbroker.com and when you login you should see the "Member Details" page. Scroll down and you will see a section where you can set-up an alias.
The alias maps to the URI that you used to register with SIP Broker.

Hope that helps
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Unread 03-31-2006, 04:42 PM   #6
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Thank you Martin and DracoFelis for that info. I'll try those ideas on the weekend.

However, it is confusing for me, that with my PSTN number mapped to 050xxxxxxxx@xxx.xxx.xxx.xxx, e164.org can call me via ENUM (test), OzTell can can call me via ENUM (WebDialer), but Sip Broker can't seem to call me via ENUM.

The ATA I'm using is the "ADSL Modem SV III", provided by NTT-West. This appears to be the standard ADSL modem they supply for their "Flets ADSL" service. It has a built-in SIP-compliant ATA (supports G.711 u-law codec only).
http://www.ntt-west.co.jp/kiki/consu...sv3/index.html
(in Japanese)

http://asahi-net.jp/en/support/guide/ip-f-sv.html
(an abridged guide in English)

The ATA is not locked - you can set it to use any SIP provider - however it is designed specifically for the bundled / packaged "IP Phone" service provided by Japanese ISPs using NTT's "ADSL Flets" connection. The dial plan is hard coded, and calls are dialed out in Japanese-domestic format.

The most flexible solution for me, would be to run Asterisk on an internal computer, have the ATA register with my Asterisk server, and then get the Asterisk server to take care of everything. However, I'm trying to see if there is a solution that would require no additional hardware, so that many other subscribers in my situation can take advantage of ENUM and VoIP peering.


edited: spelling correction

Last edited by v164; 04-01-2006 at 12:39 AM.
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Unread 04-01-2006, 07:39 AM   #7
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Quote:
Originally Posted by v164
The most flexible solution for me, would be to run Asterisk on an internal computer, have the ATA register with my Asterisk server, and then get the Asterisk server to take care of everything. However, I'm trying to see if there is a solution that would require no additional hardware, so that many other subscribers in my situation can take advantage of ENUM and VoIP peering.
I long to play with Asterisk myself, but don't currently have the time and resources to devote to it. I don't have a clear picture of your needs and what all you're dealing with, but I was able to pull most of my VoIP loose ends together using VoXaLot:

I have two DID's and an FWD number that I point to xxxxxx@voxalot.com.

I have four outbound providers for PSTN termination (one is default, the other three require dialing a prefix of #n to access them). I have StanaPhone set up as a provider so I can call other StanaPhone users. I have FWD set up as a provider to I can direct toll-free calls there.

Now, with one PAP2 registered on VoXaLot and only one phone line which is wired to all the phones in the house, I can make and receive calls to almost anywhere using the lowest cost route.

The only loose ends I have left are Gizmo Project and StanaPhone incoming calls as neither of these allow forwarding to a URI.

I'm not sure this picture will necessarily help with your situation, but like you, I hope it will give other users some food for thought.

Ron
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Unread 04-02-2006, 12:42 PM   #8
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I still can't receive an ENUM call via SipBroker. However, I noticed the modem light flashing, showing the UDP packet arriving. For some reason, my ATA was not accepting the call. To get to the bottom of the matter, I captured some of these packets and analysed the contents. Below are the SIP "INVITE" messages that arrived.


050xxxxxxxx is my 050 phone number
xxx.xxx.xxx.xxx is my public IP address
6173305xxxx is my PSTN number, which is mapped to

sip:050xxxxxxxx@xxx.xxx.xxx.xxx

at e164.org


Here are the SIP "INVITE" messages that arrived...

<< test call via e164.org - this works >>

INVITE sip:050xxxxxxxx@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 204.50.80.11:5060;branch=z9hG4bK042cc19c
From: "6044845289" <sip:6044845289@204.50.80.11>;tag=as056c135d
To: <sip:050xxxxxxxx@xxx.xxx.xxx.xxx>
Contact: <sip:6044845289@204.50.80.11>
Call-ID: 5d6...edited.out...48@204.50.80.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 02 Apr 2006 11:45:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289


I then used Voxalot Web Callback to get an incoming call via SIP Broker. For "Your Number", I entered 6173305xxxx, via SIP Broker, and for "Number to Dial", I entered 15166875089 (an ENUM test number), via SIP Broker.

<< call using SIP broker. My ATA ignores this; it doesn't ring >>

INVITE sip:050xxxxxxxx@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:24.196.79.163;ftag=as6e3737c3;lr=on>
Via: SIP/2.0/UDP 24.196.79.163;branch=z9hG4bK2618.863f3e15.0
Via: SIP/2.0/UDP 71.13.117.133:5061;branch=z9hG4bK3ad516b5;rport=50 61
From: "<my 6-digit Voxalot user ID>" <sip:blah@sipbroker.com>;tag=as6e3737c3
To: <sip:*0136173305xxxx@sipbroker.com>
Contact: <sip:blah@71.13.117.133:5061>
Call-ID: 0e3...edited.out...61d@sipbroker.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 16
Date: Sun, 02 Apr 2006 12:09:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338



The key difference seems to be in the "To: " line.

(Edited 2006-05-28: the problem is actually the "lr=on" part of the Record-Route header - see subsequent post for details)

Test call from e164.org:

To: <sip:050xxxxxxxx@xxx.xxx.xxx.xxx>


Test call from SIPBroker:

To: <sip:*0136173305xxxx@sipbroker.com>


It seems that my ATA looks at the "To: " line, says to itself "I'm not *0136173305xxxx@sipbroker.com", and ignores it.

So that seems to be the cause of the problem. But is the fault with SIP Broker, or with my ATA?

Last edited by v164; 05-28-2006 at 05:30 AM.
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Unread 04-07-2006, 05:01 PM   #9
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Hi,

The problem you mention is working as designed. In the RFC look for the section

Code:
4.1.2 Request-URI

   The  Request-URI field is a SIP URL as described in Section 2 or a
   general URI. It indicates the user or service that this request is
   being addressed to. Unlike the  To field, the  Request-URI field may
   be re-written by proxies.
Another link describes it as "bad practice"

http://www.archivum.info/serusers@ip.../msg00535.html
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Unread 04-08-2006, 02:11 PM   #10
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Thanks martin, ron, et al for your comments.

ron,

Voxalot would tie everything up nicely, except for the fact that my VSP-ISP's SIP server blocks/ignores SIP traffic from outside.

I think I'm going to have to either use Asterisk / Asterisk@Home / SER, or perhaps setup some kind of UDP proxy to pass Voxalot's signalling traffic through on the way to my VSP. Probably may as well just use Asterisk.

Although with one Asterisk box, I should be able to organise seemless ENUM lookups and/or peering for other people connected to the same ISP - kind of like replicating Voxalot's setup, but within the ISP's domain (so their SIP servers will talk to us).
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