Thanks AzRobert. Very fast and comprehensive information.
One question though. I am using Voip.ms, and their setup guide for the Obi recommends going through the local web interface instead of through the Obitalk website, and, following their recommendations, I have configured ITSP profile A, will these settings you have listed have any effect on those ones?
Bear with me, I'm still earning this stuff. One lesson I did learn though, when you're on the phone to your ISP tech support department, and they ask you to reboot your modem, dont act surprised when you call abruptly comes to an end!
D'oh!
Cheers
John
Quote from: azrobert on October 15, 2014, 08:04:12 AM
Do you have a Windows computer?
Download Phonerlite: http://phonerlite.de/download_en.htm
Click PhonerLiteSetup.exe
Take all the defaults.
Assuming:
SP4 is undefined on the OBi200
You want the outbound calls routed out SP1
Setup The softphone
Under Server Tab:
Proxy: 192.168.1.100:5063 (192.168.1.100 is the IP addr of the OBi200)
Register: Unchecked
Under User Tab:
UserName: OBi200
Click Save
OBi200
Service Providers -> ITSP Profile D -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP4 Service -> AuthUserName : (any userid)
Voice Services -> SP4 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP4 Service -> X_ServProvProfile : D
Voice Services -> SP4 Service -> X_InboundCallRoute:
{OBi200>(1xxxxxxxxxx|<1aaa>xxxxxxx|011xx.):sp1}
aaa is your local area code for 7 digit dialing.
I know this is old thread, but I can't see to get this working with current version of PhonerLite and an Obi202 at latest release.
My SIP provider is on SP2, GV on SP1. My Obi202 is at 192.168.1.120. I made the obvious changes to these instructions. (send to sp2, my Obi's ip) All I can get from the Obi is a 404 return code. (Not found) Syslog debug output from sp4 looks like this:
5/26/2016 4:05 PM,Debug,192.168.1.120,PNNCOMM:Receive sync req, set auto config
5/26/2016 4:05 PM,Debug,192.168.1.120,RxFrom:c0a8013d:5060
5/26/2016 4:05 PM,Debug,192.168.1.120,INVITE sip:15205551234@192.168.1.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK801c60040422e611bd9652ee57a91008;rport
From: <sip:blaxter@192.168.1.120>;tag=663266807
To: <sip:15205551234@192.168.1.120>
Call-ID: 801C6004-0422-E611-BD95-52EE57A91008@192.168.1.61
CSeq: 10 INVITE
Contact: <sip:blaxter@192.168.1.61:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:blaxter@192.168.1.120>
Content-Length: 230
v=0
o=- 1226971667 1 IN IP4 192.168.1.61
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.61
t=0 0
m=audio 5062 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:623333896
a=sendrecv
5/26/2016 4:05 PM,Debug,192.168.1.120,sendto c0a8013d:5060(323)
5/26/2016 4:05 PM,Debug,192.168.1.120,SIP/2.0 404 Not Found
Call-ID: 801C6004-0422-E611-BD95-52EE57A91008@192.168.1.61
CSeq: 10 INVITE
Content-Length: 0
From: <sip:blaxter@192.168.1.120>;tag=663266807
To: <sip:15205551234@192.168.1.120>
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK801c60040422e611bd9652ee57a91008;received=192.168.1.61;rport=5060
5/26/2016 4:05 PM,Debug,192.168.1.120,RxFrom:c0a8013d:5060
5/26/2016 4:05 PM,Debug,192.168.1.120,ACK sip:15205551234@192.168.1.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK801c60040422e611bd9652ee57a91008;rport
From: <sip:blaxter@192.168.1.120>;tag=663266807
To: <sip:15205551234@192.168.1.120>
Call-ID: 801C6004-0422-E611-BD95-52EE57A91008@192.168.1.61
CSeq: 10 ACK
Content-Length: 0
5/26/2016 4:05 PM,Debug,192.168.1.120,RxFrom:c0a8013d:5060
5/26/2016 4:05 PM,Debug,192.168.1.120,ACK sip:15205551234@192.168.1.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK801c60040422e611bd9652ee57a91008;rport
From: <sip:blaxter@192.168.1.120>;tag=663266807
To: <sip:15205551234@192.168.1.120>
Call-ID: 801C6004-0422-E611-BD95-52EE57A91008@192.168.1.61
CSeq: 10 ACK
Content-Length: 0
I'm not an expert on this, but is port 5060 the correct one to use? I thought based upon sp4 being 5063 that sp2 would be 5061.
As I say, I may be sending you down a rabbit hole.
The idea is to use an unused sip slot (number 4 in this case that's why port 5063) and have desktop sip phone route calls through OBI. Routing it through SIP 2 account that you use will not work.
Note that there is a very easy alternative to provide dekstop/laptop computer VoIP:
If you're using Google Voice on your OBi, then simply use Google Hangouts on your computer. If you use Google Chrome Browser, then no additional software/plugins/extensions are necessary. Chrome Browser includes built-in WebRTC support for Hangouts calling. The streamlined/basic Hangouts page is here: https://hangouts.google.com/ (https://hangouts.google.com/)
Inbound calls to your Google Voice number will ring on your OBi-attached phones, and independently, on Hangouts, as long as you leave open a browser tab logged into your Google account.
For people using a SIP VoIP ITSP that permits multiple extensions or sub-accounts, then create one, and configure it on your softphone client. I see that you are using Phonepower's OBi plan, which probably doesn't permit multiple registrations on that account, but other ITSPs like voip.ms and Callcentric can be easily configured to do this.