Two GV account. Need to ring phone and Asterisk extension using a Voice Gateway

[jcgalvezv]

I have two GV account configured in my Obi110 SP1 and SP2.

I want all my incoming calls from GoogleVoice1 (mine), GoogleVoice2 (my wife's one)  and Obitalk to ring obihai's phone and ring an extension in my Asterisk using Voice Gateway1.

I have tried several configurations but no way.

Could any experienced user please tell me, What should I configure in Gateway1 Access Number and Digimap,? and What in Obitalk/SP1/SP2 Digimap and InboundCallRoute? If something else missing please tell me what.

To test, in Phone port I configured PHONE port's OutboundCallRoute, additional to default configuration, ,{(Mgv1):gv1}, In Voice gateway1 AccessNumber I have configured 192.168.5.1:5060, in AuthUsedId the asterisk SIP user id and in AuthPassword the asterisk user password. In Speed Dial 1 I configured VG1(101). When I dial 1# in phone connected to obihai I don't see any activity in my asterisk and the slogsrv only shows "++++ lookup this _g1192.168.5.1:5060*101". If I configure in Voice gateway1 AccessNumber SP2(192.168.5.1:5060) the nothing is shown in slogsrv nor asterisk CLI.

Thanks in advance for any help.

[RonR]

With Google Voice configured on both SP1 and SP2, you cannot use Voice Gateways with SIP.  To use Voice Gateways with SIP, you must have at least one SIP provider configured on SP1/SP2.

[QBZappy]

jcgalvezv

Hi

As RonR said you will need at least one SIP account for gateway method to work.

Instead of using gateway strategy, I think you can configure the OBI

ITSP InboundCallRoute
OBiTALK InboundCallRoute
POTS-line InboundCallRoute

in the following way to make the Asterisk extension ring:
{ph,SP1 or SP2(15145551212)}

15145551212 = DID number of an Asterisk Trunk

The incoming call will ring phone and dial out over the SP1 or SP2 into your Asterisk.

[jcgalvezv]

Thank you RonR and QBZappy for your replies.

It is a pity that you need SIP in SP1 or SP2 to do voice gateways to SIP desinations. I guess this would be an interesting feature to allow SIP gateways even is SP1/SP2 are being used with XMPP.

How about IP dialing to my asterisk? Do you have any experience with IP dialing using Obi devices?

Juan C.

[RonR]

It is a pity that you need SIP in SP1 or SP2 to do voice gateways to SIP desinations.

It really is.  All that SIP support code just sitting there and can't be used.

How about IP dialing to my asterisk? Do you have any experience with IP dialing using Obi devices?

IP dialing works fine as long as you have a SIP provider configured on SP1/SP2.  Same, seemingly needless, restriction as before.

[jcgalvezv]

Thank you again RonR for your reply.

Now I see that what I was thinking to do is not doable, at least now.

I hope Obihai will consider in a near future SIP gateways, even with SP1 and SP2 using XMPP.

[GizmoChicken]

It is a pity that you need SIP in SP1 or SP2 to do voice gateways to SIP desinations.

It really is.  All that SIP support code just sitting there and can't be used.

How about IP dialing to my asterisk? Do you have any experience with IP dialing using Obi devices?

IP dialing works fine as long as you have a SIP provider configured on SP1/SP2.  Same, seemingly needless, restriction as before.

I fully agree with you both!  Hopefully this will be changed in the future!!