Configuring SIP clients (softphones) for singlestage dialing through my OBi100
Atrocia:
Hi,
I am comfortable with computers (Linux), but a telephony noob. I have my OBi100 set up with GV (SP1), and I'm trying to setup singlestage dialing (via RonR's method). I'm not sure if I've gotten everything right (some of those substitutions RonR mentions are a bit confusing, and in any event, I can't get any of my Linux SIP clients to dial out through the OBi. Most seem to need a connection to a SIP server, only some of them seem to have entries for a SIP proxy, and in most of them I can't figure out exactly how to set a userid.
With some configurations in some clients, I seem to be somehow connecting to the OBi - the phone connected to it rings. I checked the OBi call history, and those calls appear thus:
Code:
Terminal ID SP2 PHONE1
Peer Name John Doe
Peer Number john
Direction Inbound Inbound
22:14:58 Ringing
22:15:01 End Call
Can anyone provide a basic tutorial for configuring a software SIP client (on Debian stable I've tried twinkle, sflphone, empathy, ekiga, and linphone, with no luck)?
azrobert:
You are almost there with the softphone you used that got the results you posted.
RonR is making the config more complicated than it needs to be.
Remove all the changes you made except for the following:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP2 Service -> AuthUserName : (any userid)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
Replace your ITSP B X_InboundCallRoute with following:
{(john)>(1?xxxxxxxxxx|<480>xxxxxxx):sp1},{ph}
The above rule is checking the incoming call for a peer number = "john" AND a dialed number with 10 digits with or without a prefix of "1" or a 7 digit dialed number. if there is a match it routes the call out SP1. If it's a 7 digit number it will prefix it with "480" (your local area code).
If there is no match it will ring your OBI100 Phone port.
Atrocia:
Thanks much - it's working perfectly now, with twinkle! This really needs to go into a FAQ somewhere.
And just FTR, for clarity in case anyone stumbles across this thread, the reference to
Quote
Replace your ITSP B X_InboundCallRoute
is to
Voice Services -> SP2 -> X_InboundCallRoute
Atrocia:
I'm still pretty frustrated trying to get SIP clients to dial out through the OBi; here are my results so far ('userid' below is as set on the OBi, 'obi' is the hostname [or IP] of the OBi):
Twinkle - works with the following settings:
Quote
Edit -> User Profile -> User -> User name: userid
Edit -> User Profile -> SIP Server -> Registrar: [blank]
Edit -> User Profile -> SIP Server -> Outbound Proxy -> Use outbound proxy: [checked]
Edit -> User Profile -> SIP Server -> Outbound Proxy -> Outbound proxy: obi:5061
Kphone - connects with the following settings, but no sound in either direction:
Quote
File -> Identity -> User Part of SIP URL: userid
File -> Identity -> Host Part of SIP URL: obi
File -> Identity -> Outbound Proxy (optional): obi:5061
File -> Auto Register: [unchecked]
sflphone, linphone, yate-qt, empathy / telepathy-sofiasip, ekiga - could not get to connect
[Why is the Linux SIP client situation so bad - all the clients are poorly documented, untouched for years, and / or don't work very well (at least without proper documentation ...).]
azrobert:
I have Yate-qt working on Windows connected directly to GV with these parms:
Provider: none
UserName: GV_ID
Password: GV_PW
Domain: gmail.com
Server: talk.google.com
Resource: Yate
Allow plain password: Checked
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