Dektop App for using local Obi202

<< < (2/2)

mrcinaz:
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

Lavarock7:
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.

NomadTech:
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.

SteveInWA:
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/

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.

Navigation

[0] Message Index

[*] Previous page