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Thread: Italian service to route calls from a pbx to Skype

Replies: 0
Views: 7078

Italian service to route calls from a pbx to Skype 31.07.2009 16:22 Forum: Providers

Hi there!

There is an Italian service to route calls from a pbx to Skype. It costs 60 Euro per year. It could be used for example with Skypephone from H3G.
The service works fine with other IP PBX and SIP providers only with pbxes we got one way audio. They told me they need to get in touch with pbxes sysadmin in order to debug the issue.
I do not want to advertise here the service, I'm here only to ask pbxes staff to get in touch with me. Maybe the service could be interesting for other users too OR could be implemented by pbxes itself.

Thank you - pbx00


Replies: 2
Views: 13651

09.02.2008 08:46 Forum: Terminal Equipment

I'm ready to show you how it works. Please, visit pbxspace and search for pbx00. PM me your email address or search it.tlc.telefonia.voip for "Alessandro D'Arpini" to get in touch with me.
It took me a long time to discover a way to use my SPA 3000 or LinkSys 3102 as a pbxes trunk in order to dial out via PSTN or a cellphone connected to the PSTN plug of the SPA/LinkSys and, belive me, it works GREAT! :-)

See you soon


Replies: 2
Views: 13651

Daumen hoch! Guide: How to use a SPA3000 as a trunk 05.01.2008 03:40 Forum: Terminal Equipment

Please, make sure you rotate correct local UDP port to the SPA3000 / LinkSys 3102 before. In the example below the UDP port is 5064

How to use a SPA 3000 / LinkSys 3102 as a trunk:

Trunk Name: SPA3000
Language: italiano
dtmfmode: rfc2833
audio bypass: no
username 123456789
password: 123456789
SIP server: my external IP:5064 (local port of my SPA 3000 under PSTN Line
register: no
Dial Plan: 3[1-46-9]xxxxxxxx (Italian Cellphones)

Outgoing route: ItaliaCell
Trunk: SPA3000
Custom Dial Pattern: 3[1-46-9]xxxxxxxx

Now the SPA3000 / LinkSys 3102:

PSTN Line: line enable
SIP port: 5064
Proxy: leave blank
Make Call Without Reg: yes
Ans Call Without Reg: yes
Display name: leave blank
User ID: 123456789
Password: 123456789

PSTN-To-VoIP Gateway Setup:

PSTN-To-VoIP Gateway Enable: yes
PSTN Ring Thru Line 1: yes
PSTN Caller Auth Method: none

From every enabled extencion you can dial an Italian Cellphone number and the call will go out the PSTN Line of the SPA 3000 / LinkSys 3102.

You may use a DynDNS account:sipura port.
SIP server: mydomainDynDNS.net:5064 (local port of the SPA 3000 under PSTN Line)

If you have one provider enabled in PSTN Line, you have just to use those credentials under trunk SPA3000.
For example a Betamax account:
User name: Pascal
Password: Merle

Then under trunks you'll have:

Trunk Name: SPA3000
Language: italiano
dtmfmode: rfc2833
audio bypass: no
username Pascal
password: Merle
SIP server: my external IP:5064 or mydomain.dyndns.com:5064 (local UDP port of the SPA 3000 under PSTN Line
register: no
Dial Plan: 3[1-46-9]xxxxxxxx (Italian Cellphones)

Have a nice day

Alessandro D'Arpini

Thread: Getting this WiKi post back on topic...

Replies: 7
Views: 30242

01.10.2007 23:30 Forum: News

What I did with my SPA-3000 is quite simple.
Line 1 is an extension of my pbxes pbx: pbx00-extension.
All calls coming from a selection of trunks, go to pbx00-extension.
On my SPA-3000 User 1 menu I have Cfwd All Dest: mycellphonenumber@gw0.
Therefore, all incoming calls are diverted to my cellphone and the whole process is completely transparent to all callers.

In PSTN Line menu I have in the Subscriber Information menu: pbx00-another extension.
Dial Plan 3: (S0<:extension>Augenzwinkern
VoIP-To-PSTN Gateway Enabled and PSTN-To-VoIP Gateway Enabled.

When I'm on the road I can receive free calls because I attached to my Dock-N-Talk an UMTS cellphone with a convenient call plan.
When I want to make calls via VoIP while I'm on the road, I call my cellphone at home, then I dial an extension number to get in touch with friends and relatives.
As you know CDMA/UMTS won't allow to send DTMF between two CDMA/UMTS cellphones. So I did a trick. I use my Nokia E61 and Opera Mobile browser and I login to my pbxes pbx and I change PSTN number for the Classic extension (S0<:extension>Augenzwinkern.

The above scenario can give a bit of freedom to people. Next step would be the GREAT step. I mean... use the call plan in the cellphone connected to my SPA-3000 or PSTN as a Trunk of the pbxes pbx.
You have to know that the Dock-N-Talk is connected to the PSTN port of my SPA-3000. Therefore... if I have a PSTN line the scenario I talked about won't change.
You can receive calls via your PSTN or you can spare a lot of money by calling other cellphones via your cellphone connected to the Linksys/SPA device.

A GREAT way to get in touch with the world without ask for a loan, right? ;-)

I'm here, willing to help, Diafora. My device is reachable from the Internet with the right credentials ;-)

Thread: Getting this WiKi post back on topic...

Replies: 7
Views: 30242

22.09.2007 15:33 Forum: News

Hi Diafora,

I have a cell phone connected to an old SPA-3000 via a Dock-N-Talk. The SPA-3102 is a mix of the old SPA-2100 and the SPA-3000.
We have instructions on how to bridge a SPA-3xxxx to a local Asterisk but nothing about pbxes and a SPA-3xxxx behind a NAT.
I spent several nights without get it work. Therefore.... I'm here waiting for your guide!
How can we get it registered as a trunk in pbxes?


Replies: 1
Views: 10551

04.05.2007 21:31 Forum: Feature Requests

Mhmmm... An interesting question, without an answer.
Is Anybody out there with a Premium account and Joomla! installation willing to help?

Thread: Feature Request: Digital Receptionist upload

Replies: 0
Views: 6095

Feature Request: Digital Receptionist upload 24.04.2007 14:07 Forum: Feature Requests

HI there!

In my humble opinion, if we want a serious usage of the Digital Receptionist, we should allow uploads of voice prompts.
I mean, not from a local computer but from a location like, for example:

There is an issue when we upload wav files from local computers due of our internet connection. Maybe slow, maybe broken and so on.

If we could record ad polish a wav file locally, then upload it on a web server and then from pbxes web interface wrtite the URL and upload the file, maybe the whole precess would be easyear for everyone.

If that is not safe for the pbxes security policy, than we could think about FTP upload of the wav file, then point to the uploaded wav in order to get it on the Digital Receptionist tree.

Thank you very much for your patience.

Thread: RE: Full G.729 support

Replies: 33
Views: 195584

24.04.2007 13:56 Forum: News

It's strange there is only one reply to this Diafora poll.
From my side, I'll be more than happy to pay an extra fee for the G729 vocoder.

Thread: RE: Grandstream BT101 don’t register to PBXES

Replies: 1
Views: 9774

Grandstream BT101 don’t register to PBXES 24.01.2007 23:37 Forum: Terminal Equipment

I’ve a problem with my Grandstream BT101 IP phone. My phone is behind a NAT, I’m able to use it with many providers, but I’m not able to register it to PBXES. It can make a call but it cannot receive.

I suppose my case is very similar to this one: http://www4.pbxes.com/forum/thread.php?threadid=765

This is my config: http://cheatbuster.interfree.it/Varie/Gr...nfiguration.htm

And this is the connection log: http://cheatbuster.interfree.it/Varie/Log_Grandstream.htm

In the log you can see the first try to register. The server responds “100 Trying” then “401 Unauthorized”. So the phone tries another registration using the authorization credential, but in this case the server responds only “100 Trying”.

Is there any kind of incompatibility between Grandstream phones and PBXES?


Replies: 3
Views: 13020

16.01.2007 13:38 Forum: Feature Requests

You are a GREAT man!

You brought to all of us as many possibility as we can think about! At a very cheap price.
I'm not in business with VoIP. I think the VoIP as a way to be free. Not only free speech. but also Freedom to achieve the goals people need.

A small business Company, let say 5 employes, can have its own IVR, DIDs, call transfers and so on WITHOUT knowledge about Linux and without expenses for Aterisk Technicians.

I talked for hours with an Italian at the phone. He rules a small Company with interests in Turism. He was able to acheve his goals by using pbxes.
Anther guy from North Italy, with another small Company. He offers technical assistance to their customers by using pbxes.

Both of those Italians had no prior knoledge of Asterisk or IVRs or Extensions and so on.

Many other "common people" know your services and I'm sure many of them, will say Thank to You, pbxes!!!!

Quite all of them... apart competitors of course ;-)
They can continue to sell Asterisk boxes to their customers but now we can share knowledge about IP pbxes among common people.

I hope you can grow more and more because knoledge is important.

Thank You!!!!


Replies: 3
Views: 13020

Open VoIP Vs Closed One 16.01.2007 02:18 Forum: Feature Requests

Hi there!

It happens some VoIP providers, give away to their customers locked routers.
For example in Italy we have Tiscali or Telecom Italia who sells Alice VoIP. They give away SIP routers that won't allow other VoIP than their own.
Of course by using not standard UDP ports we can find a workaroud to the problem.
It's a really BIG problem, mainly because that way BIG Telcos won't allow competitors and for their customers too because they are locked and they can't choose.
Moreover... customers can't simply turn off the locked router due of the PSTN number registered with the VoIP service (they don't have POTS Line anymore) or because the locked router uses a smart card with username and password on it.
If I don't know login and password, how can I connect to the Internet by using a non locked router?

Now I'm wondering pbxes menagement if they thought about the above mentioned problem and if they have planned a way to support not standard UDP port for registration.
For example Messagenet.it uses port 5061 and yes, Tiscali users can have their ATA registred and full working behind their locked router.
Vbuzzer uses UDP port 80 and yes, it can bypass locked routers too.

I hope I was able to explain myself in my poor English.

Have a nice day.

Thread: RE: Messagenet.it - The called number answers but immediatly the line drops

Replies: 8
Views: 23113

12.01.2007 01:32 Forum: Providers


Trunks: SIP/Messagenet
Dial Rules: 021111111 or 063333333

Outbound Routing: Messagenet
Extension: 2007
Dial Patterns: 021111111 or 063333333
Trunk Sequence 0: SIP/Messagenet

I pick up the phone (ext. 2007) and I dial 021111111 or 063333333 (my home phone).
The remote phone or my home phone rings. When someone picks up the phone, the line drops.

Then I deleted Messagenet Trunk on pbxes and I used my Sipura 2100 to register with it. I called my home phone from my ATA, my home phone rung, I talked to myself (two way audio).

On it.tlc.telefonia.voip I have found Italians with same problem and tonight snom360 owner (a Premium account) called me by using his Messagenet account on pbxes. My home phone rung but when I picked up the phone, the line dropped immediatly.

We have the issue with Messagenet outgoing calls only, not incoming, nor SIP URI calls or ENUM.

On Call Monitor I have the number (063333333 for example) dialed out from Extenion 2007 with Messagenet, duration zero seconds, from-internal-cont.

Have a nice day.


Thread: RE: Messagenet.it - The called number answers but immediatly the line drops

Replies: 8
Views: 23113

Messagenet.it - The called number answers but immediatly the line drops 11.01.2007 02:30 Forum: Providers

Hi friends!

I have Messagenet.it on my Trunks and I'm able to make outgoing calls by using that Italian VoIP provider.
When the called number answers, immediatly the line drops and I get a busy tone on my phone.
It happens not only to me but to others too.

Any help?

Thank you for your support

Alessandro D'Arpini


Replies: 1
Views: 6463

Fragezeichen Dialplans 18.12.2006 00:38 Forum: Miscellaneous

Hi friends!

- I have only one extension
- I have several SIP trunks: A - B - C - D - E (five Trunks).
- In the Outbound Routing 0 I have in the Trunk Sequence = Provider A and B.
- In the Outbound Routing 1 I have in the Trunk Sequence = Provider C only
- - In the Outbound Routing 2 I have in the Trunk Sequence = Provider E only

By using provider A, B and E:

- if I dial the number 0xx. I want 00390xx.
- if I dial 00xx. the dialed number should remain as it is
- if I dial 3.xx then 00393xx. is added and it should go out by using Provider B only

As the Trunk Sequence is smart enough to try next Trunk if the first one is busy or maximum calls are reached, Provider B should go out with 0xx. and 00xx. but it should be the only one allowed to dial out numbers beginning with 3xx. (0039 added so it dials 00393xx.)

By using provider C:

- if I dial 800xx. it must go with provider C
- if I dial 90xx. the 9 shuld be stripped and only 0xx. goes out.

Can you explain this please?

Thank you very much and thank you for pbxes! It's GREAT! smile

Showing posts 1 to 14 of 14 results

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