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Is it just one line can be used when 2b is used in SP of OBi?
For CMphone, after months of use, the quality is ok. But quite often the line will drop for unknown reason. Normally 2 to 3 minutes talk are ok. But when more than 10 min, it may drop.
If the 2b credentials is used in the OBi device, after installation, you may use it as an ordinary corded phone. It means that you may have call waiting and call conference, as a result, you have a maximum of two lines.
The quality is almost the same that I am not able to distinguish either of them.
Also I am able to use to my Asterisk server to make two calls almost at the same time to make two calls to the US via the same GV account installed in SP1 of OBi110 without any problems encountered.
It works with soft phone or USB phone. According to the sale rep, they don't guarantee it will work with ATA device. Subscribers have to pay the whole year fee at the front, the only way gets money back if it won't work with soft phone or their USB phone.
Just check out from their page, here is the log of change for this update
Enhancements & Fixes in Maintenance Release 1.2.1(2289):
- Improved connectivity with Google Voice against certain routers that reboot frequently.
- Improve chances of successful NAT traversal as the OBi now processes received= and rport parameters in all final responses to SIP REGISTER, not just 2xx responses.
- Support G711A to G711U transcoding when bridging two VoIP calls.
- Fixed: OBi may reboot when it receives a mid-call SIP INFO request without a message body.
- Fixed: X_SkipCallScreening parameter does not work for anonymous incoming Google Voice calls.
- PHONE Port ChannelTxGain and ChannelRxGain parameters are applied opposing direction.
- OBi now correctly detects ring back tone from the PSTN during PSTN Connect Detection.
- Fix for SIP Remote-Party-ID header error - required for freephoneline.ca subscribers.
- Fix for out-of-band DTMF tone leakage problem
- Fix for SIP INVITE to-tag not updated properly problem causing interop an issue with Callcentric voice mail.
- Added X_ProxyRequire option under ITSP Profile - SIP.
Set this option equal to com.nortelnetworks.firewall when interop with Nortel MCS (e.g., HKBN); Users should also disable STUN and ICE when the OBi is used this way.
- Fixed *74 and *75 default value to disallow entering a 2-digit speed dial number with a leading 0.
- Added *76 to clear a speed dial
- Added Outside Dial Tone in Tone Profile
Enhancements & Fixes in Maintenance Release 1.2.1(2283):
- Google Voice calls no longer dropped when the OBi is installed behind certain home routers.
- Added Google Voice Backing Off reason on the status page.
- DSCP marking has correct default values for SIP and RTP.
- Speed dial values with extra white spaces supported.
- DHCP enhancements.
- Added more information under SP1 and SP2 Service Status for SIP:
- Show the IP address of the server that we last registered with, or currently registering with (so we know if there is a DNS error, etc.).
- Show the expiration time in seconds for the current registration.
- Show the time in seconds for the next retry if last registration has failed.
- Call back from AA fixes.
- Hostname resolution now favors DNS SRV.
- Restart not required after configuring syslog settings.
- Restart not required when setting features via star-codes.
- Use of hex values in the nonce count parameter in the Authorization header of SIP requests.
- Available codecs now ordered in accordance with priority settings in codec profile.
- Support for India PSTN Caller-ID detection (OBi110).
- #1, #2, etc. can be used as dialing prefixes for call routing.
- Added options to support NAT traversal for SIP Gateway and URL calls on SP1/2.
You may now append these URL parameters to speed dial and SIP Gateway VG1-8 access number, separated by ';',
- ui=userid[:password]
- ui=user-info, password is optional
- op=[ i ][ m ][ n ][ s ] ;option flags, i=ice, -m=symmetic-rtp, n=natted-address, s=stun
Examples:
SpeedDial = sp2(1234@sip.inum.net;ui=1000:xyz;op=sm)
VG1-8 AccessNumber = SP1(sip.inum.net;user=1000;op=imns)
Note that if userid or password is specified in VG1-8 AccessNumber, it overwrites the settings in AuthUserID, and AuthPassword in the VG.
- Improved audio quality in lossy network environments.
- Show TX and RX codec name, and tx and rx packet size for each call leg on a bridged call on the call status page.
- Improved firmware upgrade robustness to eliminate chances of corruption.
- Added Outside Dial Tone to Tone Profile A and B.
- Caller can hear the LINE port dial tone instantaneously after pressing # key on the phone.
- Call waiting and 3-way calling behavior fixes.
- Call History page now displays correctly on IE7.
發表於 2011-5-17 07:13
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