PhonePower SIP / Asterisk

[mikewis]

Ok.. I have Phonepower setup through my Obi110.  Works great but now I want to use my phonepower account as a trunk on my Freepbx/Asterisk Server.  With phonepower you can use SIP credentials but I cannot get it to work as a trunk on my Asterisk server.  Anybody use Phonepower with Asterisk? 

I register my Obi110 as a Extension on my Asterisk server but would like to use PhonePower as my outbound Trunk.

Any help appreciated

Thanks
Mike

[Taoman]

If you are subscribing to one of the PhonePower "OBi" plans you can only register an OBi device or the PP softphone. And PhonePower requires registration for an outbound call.

Edit: This string appears to register a line and make it available in Asterisk. I am unclear whether this method will work for an "OBi" PhonePower plan, however. I was fairly certain only an OBi device could register on an "OBi" PhonePower subscription.

register =>[[Auth ID]]:[SIP Password]]@[[Proxy]]/[[SIP User ID]]

Have you followed this guide?

http://www.phonepower.com/wiki/Configuring_Asterisk_To_Use_SIP_Credentials

[azrobert]

If Taoman's suggestion doesn't work, leave Phonepower defined on the OBi110 and access it from Asterisk.

Setup an Asterisk trunk without registration:
Trunk Name: OBi110
PEER Details:
type=peer
username=mikewis
host=192.168.1.110
port=5061
canreinvite=no
insecure=invite,port
qualify=yes

Host = IP address of the OBi110
Port = port number of SP2

Setup an Asterisk outbound route for the OBi110 trunk.

OBi110
Setup a dummy SIP definition on SP2 like this:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer: 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerid: Checked
Voice Services -> SP2 Service -> AuthUserName: anything
Voice Services -> SP2 Service -> X_RegisterEnable: unchecked
Voice Services -> SP2 Service -> X_ServProvProfile: B
Voice Services -> SP2 Service -> X_InboundCallRoute: {mikewis>(xx.):sp1}

Route inbound SP1 calls to the OBi110 phone port and Asterisk
Voice Services -> SP1 Service -> X_InboundCallRoute:
ph,sp2(1000@xx.xx.xx.xx:5060)

xx.xx.xx.xx:5060 is the IP address and port of Asterisk.

In Asterisk setup an inbound route with DID = 1000

[mikewis]

Well it shows Registered in PhonePower..  My freepbx shows Registered.  When I dial out on the trunk it says all circuits busy.

I did follow the Guide Taomans suggested but I am not sure what I am missing as it doesn't work.

@azrobert

So if I did what you said would I be able to get incoming calls from Phonepower on Obi and pass to asterisk?

I have GV working on Freepbx but I want to use Phonepower as it provides Caller ID name and not Just number.

Thanks
Mike

If Taoman's suggestion doesn't work, leave Phonepower defined on the OBi110 and access it from Asterisk.

Setup an Asterisk trunk without registration:
Trunk Name: OBi110
PEER Details:
type=peer
username=mikewis
host=192.168.1.110
port=5061
canreinvite=no
insecure=invite,port
qualify=yes

Host = IP address of the OBi110
Port = port number of SP2

Setup an Asterisk outbound route for the OBi110 trunk.

OBi110
Setup a dummy SIP definition on SP2 like this:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer: 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerid: Checked
Voice Services -> SP2 Service -> AuthUserName: anything
Voice Services -> SP2 Service -> X_RegisterEnable: unchecked
Voice Services -> SP2 Service -> X_ServProvProfile: B
Voice Services -> SP2 Service -> X_InboundCallRoute: {mikewis>(xx.):sp1}

Route inbound SP1 calls to the OBi110 phone port and Asterisk
Voice Services -> SP1 Service -> X_InboundCallRoute:
ph,sp2(1000@xx.xx.xx.xx:5060)

xx.xx.xx.xx:5060 is the IP address and port of Asterisk.

In Asterisk setup an inbound route with DID = 1000

[azrobert]

So if I did what you said would I be able to get incoming calls from Phonepower on Obi and pass to asterisk?

Yes, see the last 5 lines of my post.

I have GV working on Freepbx but I want to use Phonepower as it provides Caller ID name and not Just number.

You can setup Superfecta on FreePBX to get CallerID Name on your GV trunk.
I use Raspbx and had to install SuperFecta using Admin/Module Admin.
You have to configure Superfecta under Admin/CID Superfecta.
I modified the Default Scheme by unselecting everything then selecting OpenCNAM.
At the bottom of the page you have to agree to license terms and save.
You can also setup the Asterisk Phonebook and make that the primary CNAM lookup.
I also had to install the Asterisk Phonebook module.
On your inbound route for GV enable Superfecta.
The free OpenCNAM allows 10 queries per hour.
See: https://www.opencnam.com/pricing
I think using the Asterisk Phonebook will limit the number lookups you do.
In the Superfecta config there is a test function where you enter a phone number and see the result of the lookup.

[mikewis]

@azrobert

Thanks.. yes this means I dont need PP anymore.  I just tried what you said, Superfecta and works just like you said.  Great.. I too have RASPBX and it works great.  Have you created any cool scripts for Telemarketers?  This is why I setup Asterisk and to only ring the phones at certain times etc.

If I use the Obi for my ATA to my analog cordless phones and there is no Hold button, how would I put a caller on Hold? can I use a flash on the phone?

Thanks
Mike

[Taoman]

Well it shows Registered in PhonePower..  My freepbx shows Registered.  When I dial out on the trunk it says all circuits busy.

Glad you got things working. Still curious about your plan, however. Do you have an "OBi" PhonePower plan or a regular PhonePower subscription?

Q: Can I use any device on this plan?

OBihai plans are only eligible with Obihai devices.

http://www.phonepower.com/Obihai_info.aspx

[mikewis]

I have the Obihai Phone Power plan but there is an advance option to allow for SIP connection.  I have enabled that and get the SIP credentials.

But never could get it use the Phone Power Trunk.. but I had Phone Power for Callerid but now that @azrobert told me about Superfecta, I can get calleid name on GV.  So I am going to use GV instead of PP now.


-Mike

[azrobert]

I use NoMoRobo on my OBi to block telemarketers. I also have a black list on my OBi and route these calls to Zapateller on Raspbx. Zapateller is an inbound route on Raspbx that routes calls to Terminate Call/SIT Tone. Some people claim that NoMoRobo will block some legitimate calls. I have not had this problem.

I think you can setup NoMoRobo on Raspbx using a Ring Group. Raspbx also has a Black List application you can install using Module Admin. GV has a spam filtering, but I never used it.

Flash will put the call on hold, but you will hear a dial tone. The dial tone will last for 20 seconds and then you will get some nasty tones. I don't know what happens next if you continue to keep the call on hold. You can modify the dial tone to extend the time. I don't know if there is a max time period.

If someone knows how to actually put a call on hold, please chime in.