link did number -->pbxes.org-->voipcheap-->UM mobile
 

1) I have a working Eutelia voip account with a DID number: +39081XXXXXXXX
2) I have a working free account with a pbxes.org
3) I have a working voipcheap account
4) I have a working UM sim.

I want: my friends calls to lo local DID number=the UM sim ring. I want billing on my Voicheap account.

a) It's possible?
b) What step exactly to set up pbxes.org? Keep in mind: my understanding of pbxes terms like Thrunk, etc., is very poor.

thx very very much

PS. excuse me for my broken english

I will try and help you backwards! You need to first set up forwarding on voipcheap to your UM telephone number. Do you know how to do that? In the client, go to the 'Tools' menu and select 'Options'. There is a setting there, where you can enter a forwarding number. Remember to select "Always" for when you want the call to be forwarded. Any incoming SIP call that comes to your voipcheap account if you are online or not will then be diverted to your UM SIM. You will charged the standard rate. Incoming calls can come in via your SIP address which has a format like voipcheapusername@voipcheap.com.

I don't use Eutelia or pbexes but I think you can bypass pbxes.org and get Eutelia to point incoming calls from your DID directly to any SIP adress. In this case, just enter the one in the format above into your Eutelia options. (or you may need to enter SIP:voipcheapusername@voipcheap.com)

The VOIP cheap holding company blocked access to their service from pbxes.com. Check the forum on pbxes.com.

Eutelia do not consent to point to another sip provider or forward to voip address. Only consent direct forward to landliane or mobile number (UM mobile also of course) but eutelia rate for the mobile are very high (from 18 to 30 eurocent per minute).

Ok. Thx :(

Is there another possible solution for link my free italian DID number (Eutelia or messagenet provoder) with a voipchip or another cheap betamax provider? :(:down:

Thx in advance

Voxalot should work
 

That's correct. Voxalot.com, however, should work. Unlike pbxes, they seem to have an agreement with Betamax. Create an account there, you'll have to pay a tiny annual fee for inbound calls registered through Skypho/Eutelia, and then set up a forwarding to your UM mobile over your outbound Voipcheap account.

I use voxalot on a daily basis and can wholeheartedly recommend it.

umhhh, ok i'll try it
Thx :rolleyes:

betamax was blocked some time ago at pbxes but not anymore. I am using internetcalls a betamax service without any problems at pbxes.

I was also using justvoip.com using their forward and than have pbxes call username@sip.justvoip.com This worked fine till several month ago when the call quality became bad, really bad.
When forward is off and my softphone is on I can have pbxes call my justvoip account and call quality is ok.

The other solution you could use is a sip based betamax solution. Justvoip like voipcheap is a softphone based only solution and does not support sip calling.

http://backsla.sh/betamax to find different betamax solutions and how much it cost. www.freecall.com would be an option and 10 eurocent per minute to um+ numbers.

with UM I suspect you mean their new +44 based number and not the lichtenstein as lichtenstein can't be reached using betamax.

To setup bpxes, you go to trunks to register your DID number
also register your betamax sip account here.

next go to inbound routing.
add incoming route
under trunk you type the name of your italian did trunk
also select ring group1
submit and apply

next go to outbound routing
add route
give a route name
dial pattern: XX.
Trunksequence: select your betamax provider here (the one you added under trunks)

go to extensions
create classic extension
give an extension numer, for example 100
under pstn number, type your um numer (example 00447etc)
submit and never foget apply (red bar on the top)

go to ring group1
add extension 100 (or other number you gave, see above)

submit

now you are ready.

Incoming call will now come in from thru your did trunk, via incoming route they will ring on the extension you added under the ring group.
your extension is a pstn extension which means that when it rings it will trigger the extension to go to outbound route, trunks (betamax) to call your mobile.

I have old UM lichtenstein number, and i'll upgrade in future to UM+. I haven't try to call lichtenstein mobile from voipcheap but in the rate table of voipcheap lichtenstein mobile is present (and i imagine is reacheble).
http://www.voipcheap.com/en/rates.html

Thx very much for the great aid bbob:grin:

bbob, that was very helpful. Until you explained it, I just couldn't see what pbxes was all about.

I have a less elegant solution for using Voipcheap, which is only 8 eurocents to UM+. My regular voip provider, Lingo, has a feature where you can forward calls to any pre-determined number by hitting "0" when you are in voicemail. I just have that pointed to the access number for Voipcheap's "mobile" service. I can then key in the UM+ number at that point. However, it's not something I would ask callers to do. For that, the pbxes/DID solution looks much better.

I still have a lichtenstein UM number, but in the past calling the number using a betamax solution was not possible, also the rates to lichtenstein are to high. Whenever I tried calling the number I sometime got so eastern european speaking person on the phone, next time it would ring but my phone would not ring. For some reason they have problems with lichtenstein mobile.

I have now switched to UM+ and calling +44 is no problem for betamax and you can get low rates, average 10 cents per minute.

pbxes is a very strong solution which offers many possibilities. The only problem is that it is a do it yourself solution, which does not make it plug and play.
The is a forum at pbxes but you can only post if you are a paying user (cost 11 euro per month).

The free version has most features activated only difference with the paid version is, only 2 calls are allowed at the same time, no callback, no fax receiption and some other small thing.

But if you get pbxes to work, it's a great solution.

In my situation I use it as my office and home pbx. I have an office in both Holland and Belgium.
I have several DID voip numbers for different countries. Whenever I am at home/office the DID numbers will ring on my normal phone (voip phone). Parallel every call also rings on my mobile (takes about 10 seconds longer to ring tahn the voip phone).
So when in the office I can pickup the phone answer the call. When traveling I can pickup the call with my callphone.

Basicly this is what Yackie and Maxroam are now trying to offer. The difference is that their solution is more expensive and you pay a high price for DID numbers.
Their advantage is that they do all the work for you, forwarding etc, but their system does not offer the option that pbxes has.

I have the same experience with Um mobiles and betamax. Calls to Um did not get through and sometimes I was charged despite the call not being connected.

bbob, I tried just now to set this up using IPKall as the DID. Perhaps I didn't do it correctly, but I think it didn't work because IPKall does not offer a true SIP address -- I was using the IP or URL address that the IPKall email said calls would be coming from (voiper.ipkall.com). I could see from the status page that pbxes had logged into VoipCheap, but the IPKall trunk was grayed out.

Do you know whether IPKall numbers can be used as DIDs for pbxes? Alternatively, do you know of any free SIP DIDs?

ipkall can be forwarded only to a sip account.

What you can do is open an account at freeworld dialup (fwd) and forward the ipkall numer to that account.

under trunks you than register the fwd account and it should work.

pbxes can somehow also be used to call direct username@pbxes.com or something. I have not tried this but you might want to search the forum at pbxes.

As for the USA, I believe that there are no free DID numbers anymore. Some charge between 2-3$ per month for a number. Ipkall seems to be the last one.

It's the other way round: to use IPKall DID you have to have SIP account somewhere else the inbound calls can be forwarded to. A solution which can surely work is to register an account at eg. draytel.org and then use your Draytel account number for "SIP phone number" and draytel.org for "SIP proxy" field. So a call made to your DID will appear as coming from the draytel.org. You then register your draytel.org credentials into pbxes and it should work. You can use any other VoIP provider that peers with ipkall.com though, the only thing I'm not sure whether the IPKall DID is restricted to SIP accounts with numeric usernames.

My regular landline phone, Lingo, is a SIP number, but I of course don't want all my regular calls going to UM+. Is there a way to forward the IPKall number to my Lingo account, register the Lingo account with pbxes, but then have pbxes divert to UM+ only those Lingo calls that come in on the IPKall line? I'm going to end up with so many accounts I won't be able to keep track of them. :)

No that will not work, it's best te keep a seperate account to forward the IPKall number and have that seperate account registered at PBXes.

Although if you have lingo attached to pbxes and forward ipkall to lingo all lingo calls will be forwarded to your UM+
But you could also login a normal voip phone into the pbxes account.

This means that both lingo and ipkall calls will ring on your voip phone and after 10 seconds on your mobile um+

So when at home you can just answer the phone on your voip and when not at home it will also ring on um+ and you can answer it there.

but if you life in the USa you don't want to answer a um+ phone as the cost for incoming calls will be high.

I remembered I had an old FWD account, so I set it up with that. I'm getting a busy signal, but I think I've made some progress because the call monitor does show the call trying to get to the right place. Here is the relevant portion of the monitor log (I hope it formats ok in this window):

Date Time Caller ID Number Destination Trunk Context Duration

2007-10-24 13:02:06 00447937xxxxxx from-internal-cont 0 sec
2007-10-24 13:02:01 00447937xxxxxx from-internal-cont 0 sec
2007-10-24 13:02:00 "my location" "my cell no" 100 fwd.pulver.com ext-local 0 sec


It has the correct caller ID, extension (100), and trunk on the incoming leg. It does not show a trunk on the outgoing leg, but it does correctly identify the destination telephone number. Of course, it's always possible that this is set up correctly, but I'm getting a busy signal because of some problem with UM. When calls go through, I should get the UM voicemail announcement.

How does this look to you?

Edit: I've checked it over a period of time now, and it's always busy. When I dial the number directly (but using VoipCheap), I get through to the UM voicemail announcement. So something in pbxes is not working right. I'll look through the forums and see what I can find. I sense that it's very close -- perhaps something as simple as number format or dial rules.

look at your ring group !

You have an incoming route and in the incoming route you can set this route to ring on a specific extension or on a ring group.

Try ring group first.
Add an normal sip extension and register a voip phone to that extension.
Put this extensiob number in the ring group and also have the incoming route set to this ring group.
Try to call now and see if the phone rings.
If so the next step is to add the pstn extension (which calles your um number)

bbob, thanks for all your suggestions, but I still cannot get this to work with PBXes. I have confirmed that each leg of the connection is working:

I can dial my IPKall number from any phone, and it rings in to my FWD account using the FWD softphone. Audio is fine.

I can dial in and talk from my FWD softphone to my VoipCheap softphone using my VoipCheap SIP address.

I can dial out and talk from my VoipCheap softphone to any phone, including my UM+ number.

However, when I try to do this through PBXes, I get a busy signal. This happens even when the extension is my home PSTN number or my home VoIP number.

My PBXes configuration is as follows:

Classical extension 100 is my home PSTN number (for testing); all other fields are the defaults.

Ring group 1 uses extension 100. Ring strategy is ringall, ring time is 60. All other fields are the defaults.

Two SIP trunks: FWD and VoipCheap. The General Settings for both are the defaults. Account information is correct (username, password, sip server), as indictated by the Status page, which has grayed icons when the information is incorrect. The FWD trunk registers for both inbound and outbound calls, the VoipCheap trunk is also set for both inbound and outbound, although I have tried it both ways. No dial rules or other optional settings for either trunk.

The Inbound Route uses the FWD trunk, with the Destination Ring Group #1. All other fields are the defaults.

Outbound Route 0 has the name Outbound, Dial Pattern XX, Trunk Sequence SIP/VoipCheap. All other fields are the defaults.

All other settings are the defaults.

I've tried variations on this configuration, but usually I just get a busy signal. On the inbound route, you can set the destination as an extension, a ring group, or Callthru. I've tried all of them. I believe I got a dial tone rather than a busy signal when I used Callthru, but I couldn't dial anything from that dial tone. On the Incoming Route page, you need to enter the name of the trunk rather than select it from a drop-down list, so it's possible I didn't identify it properly. However, I also tried the FWD username in identifying the trunk, and that didn't work either. On the Trunk page for FWD, it says that the trunk is not used by any routes. However, since that statement appears to refer to OUTBOUND routes, I didn't think it was a problem.

The best hint about what is going wrong is probably on the Call Monitor page. The incoming call is usually correctly identified. Depending on the configuration, the destination might be identified as "s" (whatever that means), or 1 (for Ring Group 1), or 100 (for Extension 100), or it might give the number of the extension. The only trunk that is identified in that log is fwd.pulver.com. I do not see sip.voipcheap.com anywhere on the log, even though PBXes appears to be registering properly -- the Status page shows the trunk as lit, and if I put in an incorrect password, it goes gray.

I've done a lot of troubleshooting, but at this point I'm stumped. Any advice would be appreciated.

I found the problem. The Dial Pattern in Outbound Routes needs a period, i.e., XX. rather than XX :rolleyes:

But it works great now. Thanks for all your help.

great you solved it.

My advice was make a sip extension 101, have a sip softphone conect to extension 101. add 101 to ring group.

Now you have this working you can still do this.

This means you would have extension 100 which is a pstn extension which forwards to your mobile.
You have extension 101 which is a sip extension and you have a softphone or hardware sip device at home connect to this extension.

When at home you can just answer the call on your normal sip connected phone, when away you pickup your mobile.

Great idea, but I wanted to set this up so that I and others could call my wife, who is traveling and has the UM+ card. I don't want it to ring my phone every time she gets a call. In fact, at this point I'm not even directing the call to her cell, but to the landline at the apartment where she is staying.

I did not see a way from the free account to put in several extensions and give the caller a choice of which one to ring, e.g., with an outgoing message telling the caller to press one for me, two for her apartment, three for her cell, etc. Can that be done with pbxes?

It can be done with pbxes but you need the pro version.

In the pro version you can setup a complete voice response system (digital receptionist) and have the caller press a button to make a selection, this can be connect to an extension or to voicemail.

You can make different menus, so you can say did line 1 is going to menu nr 1 where the caller can press 1 to get connected to you or 2 for your girlfriend.

On did line nr 2 you can have the caller select between your home or your mobile.

When they call you digital receptionist will answer the phone straight away.

Pbxex Pro sounds great, but at 10 Euros per extension per month, I can't justify it. I'm doing this more for sport than because I need it.

I did try to set up a second inbound trunk (with a free sipdigits/clickdigits number). Pbxes registered the trunk, but I couldn't get it to work. I did it exactly the same way as the FWD trunk, trying both the same Ring Group and a second Ring Group/extension. I suspect that either the free pbxes account doesn't allow a second trunk or sipdigits is somehow blocking.

I like pbxes, but there is not really a lot of information on how to use it.

my mistake, pbxes premium will work, also cost 10 euro per month. pbxes pro add lots of things for keeping track of sales, phones etc = total office solution which you don't need.

I have premium myself and use digital receptionist also, including a small menu where people can leave a voice mail.

I'm sure you know this already, bbob, but I discovered that an incoming route linked to Callthru allows the caller to dial either an extension or an external number, which will then go out over the VoipCheap trunk. Of course I needed to get an additional DID in order to do this and still allow dialing in to my wife's apartment in Europe over the other DID.

Amazing that all this can be done essentially for free!

I've been playing with this FWD -> pbxes -> Voipcheap combination over that last few days, and it occurs to me that it may provide a good solution for outbound calling as well as inbound calling. Is there any reason why you couldn't purchase a local sims card, and then call the SIP Broker access number for that country, enter the SIP code for FWD, and get a dial tone from Voipcheap using pbxes's Callthru feature? It's a lot of numbers to key in I know (although speed dial helps), but you could actually call the U.S. and about 35 other countries for the cost of a local cellular call. AND, your friends could call you with a local number as well -- either your DID number or a SIP broker number. I've tried it with a couple of SIP broker numbers, and it seems to work fine.

Is there enough difference in the rates and costs of local sim cards versus international sim cards that it makes sense to do something like this?

I haven't done all this, but here's something to read

http://scopezoom.com/guide6.htm

Yes, that's the same idea, but with the ability to choose different outbound trunks based on the dialplan. The one disadvantage of using a different SIM card for each country, compared to a single international SIM card, is that you would need to redirect the inbound PBXes trunk whenever you changed cards. This would be invisible to callers, but it means finding an internet connection every time you change location, which is something of a pain.

After lots of trial and error, I did finally figure out how to remotely change call forwarding on the free version of PBXes, i.e., to get call following. So this would make it possible to give friends a single US number (or whatever other DIDs you obtain), and to use that number to reach whatever SIM card you put in the phone. Both incoming and outgoing calls would be at VoIP rates, plus the cost of a local call on a local SIM chip.

looks great but very complicated.

I would still prefer pbxes using callback.

Reason is simple. If you buy local prepaid cards you think that you get a good deal, but for example here in europe most simple prepaid cards do charge a minimum of 0.20 euro per minute.

If you would have a united-mobile card you can make a callback using betamax for around 0.10 to 0.15 euro per minute. When doing this in roaming free countries it's cheaper than a prepaid card.

When in countries where the there is no free roaming you could indeed use callthru.

This just saves you buying all different sim cards and keeping track of them.

It's possible to do that without pbxes. Just point the DID to the SIP address of the provider chosen to forward the call.

There are an increasing number of mnvo providers with cheap int'l calls from a few cents.

But if they don't list good rates for all countries, or for calls to the same country, then callthrough via a foreign access is worth looking at.

It may indeed be that callback has similar rates, but sometimes just pressing one or two buttons to dial a saved entry seems a bit easier

I don't intend to actually use it; I'd just like to figure it out. As someone said in another forum, if I count my time, I've got my costs down to only a couple of hundred bucks per minute. ;)

Seriously, when you say you prefer "pbxes using callback," are you referring to the callback feature of pbxes? If so, I don't believe that is available in the free version.

In any event, what I'm proposing does not preclude using United Mobile or any other card. I can set up pbxes so that one DID rings whatever cards and/or landlines I choose, a second DID allows me to change those target numbers remotely, and a third DID provides callthrough dialtone using any of several cheap outbound trunks (maybe there is a way to do this with fewer DIDs, but I haven't got that far yet). The only cost would be the outbound trunk, which is often free to landlines, plus the cellular connection which is often free inbound. But the system would work the same for both international and local cards.

I do agree with your point about buying and keeping track of lots of different cards. Any savings are likely to evaporate once the minutes expire.

I tried this with IPKall and VoipCheap, but for some reason the forwarding didn't work. A call to the DID rang the VoipCheap softphone, but, if I recall correctly, I always got a busy signal when I tried to forward the VoipCheap line to another number.

I wonder if it's more to do with IPKall settings, as this works fine with UK DIDs I use. And if the softphone is online it will ring both this and the forwarded destination, the latter starting a few seconds later. Have you tried Freedigits?

What I do is create an extension on my handset. I've got it programmed into Fring and into the Nokia SIP phone. Fring works ok over EDGE. Using either wifi or EDGE I program in a co nditional call forward once I get the SIM. EDGE is usually over a contract SIM for two minutes before I swap SIM cards.

IPKall offers a free DID, and nothing else -- i.e., there is nothing on its website for sale, no customer service, etc. I think it then just forwards the call. There is very little to configure. Perhaps VoipCheap doesn't like double-forwarding from IPKall. I know there was a period where the Finarea companies were blocking pbxes, apparently because a pbxes subscriber tried to set up a callback business using pbxes and a Finarea company.

Anyway, I don't mind using pbxes to do this. It seems to work just fine, and it offers some flexibility. For example, you could set up a Ring Group so that the DID would ring all of your SIM cards simultaneously. Then you wouldn't have to bother with remote forwarding.

I do have a Freedigits account, but I also use it with pbxes. It's the DID I use for remote call forwarding.