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Unread 04-29-2006, 11:09 PM   #1
pmerrill
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Default SIP to SIP Calls and e164.org

A quick question to see if my understanding is correct. I have put my home PSTN number on e164, so presumably if anyone makes a call to my PSTN via VoIP, the call will be free if their provider looks the number up on e164 first.

The process, as I understand it, is that my home pstn 9452XXXX would get translated to my sipme number 1777XXXXXX and the call would be placed directly to the 1777XXXXXX number for a cost of $0 (in most cases depending on their VSP provider).

I've tried to test my home number with voxalot (which I thought did the e164 lookup) and was expected it to route the call directly to 1777XXXXXX@sip.sipme.com.au but instead it routed it to 029452XXXX@sip.sipme.com.au.

Is this correct?

Another thing I don't quite understand. From reading about sipbroker, and given that I have a DrayTek 2100V, it appears that I have to route my calls to sipbroker first so that it can sort out where to send the outbound call. But since I have a DID provided by sipme, doesn't that mean that sipme would be unable to route calls to my DID (especially from the PSTN network)? So if someone phoned my DID from the PSTN, there is no way SIPME could route the call to me unless they too used sipbroker on the inbound call? So,

Call to 02XXXXXXXX (my DID) from someone on the PSTN
PSTN routes call to SIPME exchange
SIPME looks up 02XXXXXXXX on sipbroker and finds there is a match to 1777XXXXXX@sip.sipme.com.au
SIPME routes call to 1777XXXXXX@sip.sipme.com.au
***but since my ATA is not registered with sipme but with sipbroker, then sipme thinks I'm off-line and fails the call???

The only way that I can see this all working is for the VSP (sipme in my case) to use sipbroker in their routing process.

Am I on the right track or completely in left field?

Last edited by pmerrill; 04-29-2006 at 11:46 PM.
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Unread 04-30-2006, 08:16 AM   #2
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Quote:
Originally Posted by pmerrill
I've tried to test my home number with voxalot (which I thought did the e164 lookup) and was expected it to route the call directly to 1777XXXXXX@sip.sipme.com.au but instead it routed it to 029452XXXX@sip.sipme.com.au.
You have to instruct VoXaLot to change the number to International format before sending to ENUM Lookup. You can do that in Dial Plans, Advance Mode. International format is 6129452XXXX It will not affect the number sent to Sipme.

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Unread 04-30-2006, 09:14 PM   #3
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Thanks for the info, it seems to work.

However, just for my info, in order to take advantage of e164, do I have to do it by using services like sipbroker or voxalot or do most VSPs use the service so that I don't have to?
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Unread 04-30-2006, 10:15 PM   #4
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Quote:
Originally Posted by pmerrill
However, just for my info, in order to take advantage of e164, do I have to do it by using services like sipbroker or voxalot or do most VSPs use the service so that I don't have to?
Pretty much any VoIP provider could add ENUM to their service, if they chose to do so. But many still don't for some reason (although the numbers that do use ENUM are slowing rising). So for the time being, using some ENUM enabled service (such as SIP Broker or VoXaLot) is the easiest way for most users to dial e164.org numbers.

NOTE:
It's possible to get "customer owned" equipment that will do e164.org/ENUM lookups directly, without going through any VoIP service to dial ENUM. For example, the open source http://www.asterisk.org IP-PBX, has direct "built-in" support for ENUM. So with the right equipment, you can do the ENUM lookups directly (and then just have your VoIP equipment make the direct "peer to peer" call shown in the ENUM database). It's just that most "end users" find it easier to use a service like VoXaLot (to make the ENUM calls), than to have to get/use fully ENUM enabled VoIP equipment...
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Unread 04-30-2006, 11:40 PM   #5
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Quote:
Originally Posted by DracoFelis
Pretty much any VoIP provider could add ENUM to their service, if they chose to do so. But many still don't for some reason (although the numbers that do use ENUM are slowing rising). So for the time being, using some ENUM enabled service (such as SIP Broker or VoXaLot) is the easiest way for most users to dial e164.org numbers.

NOTE:
It's possible to get "customer owned" equipment that will do e164.org/ENUM lookups directly, without going through any VoIP service to dial ENUM. For example, the open source http://www.asterisk.org IP-PBX, has direct "built-in" support for ENUM. So with the right equipment, you can do the ENUM lookups directly (and then just have your VoIP equipment make the direct "peer to peer" call shown in the ENUM database). It's just that most "end users" find it easier to use a service like VoXaLot (to make the ENUM calls), than to have to get/use fully ENUM enabled VoIP equipment...
Thanks for the info. Unfortunately, the problem of inbound DID always appears then. In order to use Voxalot I need to register with Voxalot and then SIPME cannot route calls made to my DID. What I need to somehow do is register with SIPME and route calls to Voxalot, which then can route calls back to the "best" provider, which could include SIPME. Unfortunately, that means that Voxalot would need to provide some other means of identifying me other than me registering with them.

Asterisk is a great solution but I don't want to run a PC all day and night. That's why my router's there.

I guess I'll wait until the industry matures a bit and these problems sort themselves out.
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Unread 05-01-2006, 01:37 AM   #6
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Quote:
Originally Posted by pmerrill
Unfortunately, that means that Voxalot would need to provide some other means of identifying me other than me registering with them.
VoXaLot does not require your ATA to be registered with our proxy. The ATA can use VoXaLot as a gateway (if your ATA supports this).

Just means that the your VoXaLot inbounds will either goto voicemail or use call forwarding (if set-up).
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Unread 05-01-2006, 07:26 AM   #7
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Quote:
Originally Posted by martin
VoXaLot does not require your ATA to be registered with our proxy. The ATA can use VoXaLot as a gateway (if your ATA supports this).

Just means that the your VoXaLot inbounds will either goto voicemail or use call forwarding (if set-up).
I believe I understand what you are saying. Unfortunately, the Draytek 2100V does not support gateways. How could I confirm this? I believe that I need to register with SIPME so that they can route the call to my ATA (i.e., username and password). The Draytek 2100V does have an Outbound Proxy setting but if I set this up to point to Voxalot, then how will Voxalot know which user's dialplan's to use? The 2100V setting are as follows:

SIP
SIP Port : 5060
Domain : sip.sipme.com.au
Proxy : sip.sipme.com.au
Outbound Proxy : sip.sipme.com.au
Stun Server :

Ports Setting
Port 1
Use Registrar check box
Display Name : SIPME
Account Name : 1777XXXXXX
Authorization User : 1777XXXXXX
Password : ********
Expiry Time : 1 hour

Would it work if I changed the Outboard Proxy to voxalot.com? Is there any combination of setting that would work?
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Unread 05-01-2006, 08:33 PM   #8
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Quote:
Originally Posted by pmerrill
I believe I understand what you are saying. Unfortunately, the Draytek 2100V does not support gateways.
I believe you are correct, however it does support direct SIP peering for 60 friends. 8.2.1 in the manual

Quote:
I believe that I need to register with SIPME so that they can route the call to my ATA (i.e., username and password).
This could be correct, however SIPME support peering. Find out if you can SIP forward all your calls to another SIP URI. If you can, forward all your calls to the VoXaLot account and then register your ATA with VoXaLot. Read this tutorial http://www.voxalot.com/action/static...lay&itemOID=36

Quote:
The Draytek 2100V does have an Outbound Proxy setting but if I set this up to point to Voxalot, then how will Voxalot know which user's dialplan's to use?
You will need to set up VoXaLot dial plans at VoXaLot. It's not hard. Have a look at this tutorial. http://www.voxalot.com/action/static...lay&itemOID=23

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Unread 05-01-2006, 11:01 PM   #9
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"I believe that I need to register with SIPME so that they can route the call to my ATA (i.e., username and password)."
I have Sipme and my ATA is registered to Voxalot and Sipme is set up as provider in Voxalot with Voxalot dial plans only - this works - bill
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Unread 05-02-2006, 02:10 AM   #10
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Quote:
Originally Posted by wildbill
I have Sipme and my ATA is registered to Voxalot and Sipme is set up as provider in Voxalot with Voxalot dial plans only - this works - bill
I had that working too and it works fine but does not solve my problem of inbound calls from a DID (I think).

For example, let's assume that my DID is 02 9451 2222. This is a number that I can give to people who don't use VoIP but regular PSTN telephones. This number has been provided to me by SIPME for $5 per month.

When they phone this number, Telstra switches the call to SIPME who then lookup that number in their database and find that it maps to 1777XXXXXX@sip.sipme.com.au (my VoIP number). SIPME then route that call to my ATA and my telephone rings. The only problem is that since I have registered my ATA with Voxalot, SIPME think that I have my ATA turned off and they don't have my TCPIP address, so they can't send the call to me. Net result is that my phone does not ring.

One solution I know of is that if SIPME look my number up with sipbroker, then they will find that 02 9451 2222 is really 8XXXXX@voxalot.com. SIPME could then route the call to voxalot who could then route the call to my ATA because I have registered with them and thus they know my TCPIP address.

I don't believe SIPME do this.

The other option is for me to request SIPME to point the DID provided to me to 8XXXXXX@voxalot so that when someone rings 02 9451 2222, the call gets routed to Voxalot, they see that I'm connected because I've registered with them, route the call to me and I hear my phone ring.

I don't believe that SIPME will do this either, even though it doesn't really cost them anything. They get paid by me for outgoing calls not incoming calls.

Sorry for being so specific but in trying to find a way around this problem I've run across some suggestions people have provided that are wrong, perhaps from them not fully understanding the problem.

If I am completely wrong here, then someone PLEASE let me know.
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