Using CSipSimple With OBi
dinlaca:
A couple of more observations I have:
If you add a filter of "Directly Call" and "All", then you won't be prompted in native phone dialer to choose between Mobile Phone and SIP account; all calls will be made through the SIP account.
Also, re: the version of Csipsimple which works, I too initially had difficulties getting things to work and stay working with the newer versions. But, I set up the SIP plan/account with the 1899 version available at nightlies.csipsimple.com/trunk, and then upgraded directly to the 2025 version available there, and have not had a problem since the upgrade (other than sound quality, but I am still working on that codec).
Thanks so much for having motivated me to integrate my cell use (through data plan) into a very functional Google Voice plan through my OBi110.
Now, the last step which would make this PERFECT would be to find a solution that permits both outgoing and incoming calls over the cell phone through cell data plan without use of a third party sip provider (like sip2sip.info); it would have one less step (and one less company to worry about going under), and that much more long term reliability. But, I will leave that for better minds than me to figure out.
ianobi:
dinlaca,
Good to see your setup working so quickly. It took me longer than that and I’m still fine tuning!
Your observations on CSipSimple are interesting. I may try an upgrade later using your suggestion.
For further investigations, it is worth looking at your account web page on sip2sip. “History” has a really good SIP debug section to show exactly what happens to a call. When in the debug mode for a call hit the “media” button to show codecs etc.
I’m sure that you have realised that the sip2sip account on your sp2 is there for outbound calls and is not needed for the inbound calls to the OBi. However, it can be used as just another SIP service for any calls coming into sp2.
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Now, the last step which would make this PERFECT would be to find a solution that permits both outgoing and incoming calls over the cell phone through cell data plan without use of a third party sip provider (like sip2sip.info)
I have thought about this, but never managed to make it work! It’s quite easy to set this up using a softphone or ATA where you know the IP addresses. It requires using peer to peer calling without registration. There are examples on this forum. When on home wifi, I can call the android phone at its wifi address using something like sp2(anything@192.168.1.12). I cannot seem to call the other way. This needs more testing – any volunteers? :)
azrobert:
said by ianobi:
Quote
Now, the last step which would make this PERFECT would be to find a solution that permits both outgoing and incoming calls over the cell phone through cell data plan without use of a third party sip provider (like sip2sip.info)
I have thought about this, but never managed to make it work! It’s quite easy to set this up using a softphone or ATA where you know the IP addresses. It requires using peer to peer calling without registration. There are examples on this forum. When on home wifi, I can call the android phone at its wifi address using something like sp2(anything@192.168.1.12). I cannot seem to call the other way. This needs more testing – any volunteers?
All that is needed is a softphone app that doesn't require registration.
The only one I found is Mizudroid, but it doesn't send username with no registration. Don't know if this is a bug or intentional.
Do you know of any softphone apps that don't require registration?
Update:
My above post isn't true.
You would only have outbound calls from your handset.
Something else would be needed for inbound.
hwittenb:
Quote from: azrobert on November 30, 2012, 09:06:19 am
All that is needed is a softphone app that doesn't require registration.
The only one I found is Mizudroid, but it doesn't send username with no registration. Don't know if this is a bug or intentional.
Do you know of any softphone apps that don't require registration?
The mobile phone softphone app Acrobits (plus their enhanced softphone Groundwire) has an configuration option for a sip account to not register. In the settings it is a check mark to set the account for outgoing calls only. The softphone allows you to setup multiple voip accounts so you can have one account like this and a different account that registers for incoming calls.
This is a report of my Acrobits test today. I setup a direct account on Acrobits to send dialed digits to my OBi110. I setup the account with my OBi's DynDNS symbolic address and my SP2 sip port as the account proxy server and I set it not to register. The sip port number is set to forward in my router to the OBi. This should allow single stage outbound dialing thru the OBi. On SP2 I setup an inbound routing element to send the dialed digits to SP1 based on the incoming caller. This will bridge the call out thru Google Voice. Calls seemed to work well except for dtmf digits transmitted over the in-progress call after the call was connected. Maybe the dtmf problem was due to my LG mobile phone.
I have SP2 setup to register to a voip provider. When I altered the inbound routing on the OBi to try to bridge the call back out thru SP2 or a VGx tied to SP2 there were audio problems and this technique was not satisfactory. Altering the inbound routing on the OBi to bridge the call on the Line port was OK except for the dtmf problem mentioned above.
ianobi:
Interesting tests! It would seem that direct peer to peer calling is possible from cell phone to OBi as hwittenb suggests. This does give a free link to the OBi and all its services without a third party being involved. Calls from OBi to a cell phone are always going to need a third party that is registered, as the cell phone IP address will be unknown - except as i suggested if you are at home on home wifi.
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When I altered the inbound routing on the OBi to try to bridge the call back out thru SP2 or a VGx tied to SP2 there were audio problems and this technique was not satisfactory.
I changed this in all my trunks and it seems to solve some problems:
Voice Services > SPX Service > MaxSessions: 4
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