Thread: Italian service to route calls from a pbx to Skype |
|
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
|
|
Thread: |
|
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
|
|
Thread: |
|
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.
Example:
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... |
|
22.09.2007 16:33 |
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?
|
|
Thread: |
|
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 |
|
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:
http://64.191.6.75/test.wav
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 |
pbx
Replies: |
33 |
Views: |
231540 |
|
|
24.04.2007 14:56 |
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: |
|
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!!!!
|
|
Thread: |
|
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 |
|
Hi!
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.
Alessandro
|
|
Thread: RE: Messagenet.it - The called number answers but immediatly the line drops |
|
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
|
|
Thread: |
|
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!
|
|
|