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Author Topic: Release Notes - OBi100 and OBi110 ATAs  (Read 73615 times)
ShermanObi
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Posts: 177


« on: February 01, 2012, 06:18:52 pm »



OBi100 and OBi110 Release Notes

Firmware release 2872 is a major release with updated support for the new Google Voice OAUTH authenication method

This firmware is applied to your device via the OBiTALK portal.  To upgrade your device ensure it has been added to the OBiTALK portal.  If Google Voice is already configured on your device it should prompt you to upgrade.  Or if you add Google Voice to your device it will also prompt you to upgrade.

These notes detail all significant features and bug fixes implemented since the previous release noted in the post below:

Device Enhancements:

  • Various Google Voice enhancements including a new authentication method and compatibility improvements
  • Fixes to xml parsing method for user-developed xml configuration files
  • Various minor feature changes for some service providers

21-Nov-14 Correction - this post previously incorrectly referenced firmware version 2877 instead of 2872
« Last Edit: November 27, 2014, 08:06:38 am by MarkObihai » Logged
MarkObihai
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« Reply #1 on: November 14, 2014, 09:03:26 am »

Archived Release Notes:

1.3.0 (2824)
Various bug fixes and performance enhancements.

1.3.0 (2774)
- Fix for playing out-of-band DTMF received via SIP INFO method.
- DNS SRV (TCP) Fragmentation Enhancement.
- Various software optimizations to improve performance.

1.3.0(2765)
- Fix for playing out-of-band DTMF received via SIP INFO method.
- DNS SRV (TCP) Fragmentation Enhancement.
- Various software optimizations to improve performance.

1.3.0 (2744)
- Various software optimizations to improve performance.
- Support for fax pass-through event (RFC2833).
- Show fax mode active state in call status.
- Proxy redundancy fail-over then resume improvement.

1.3.0(2721) for OBi1 Series:
- Housekeeping Release: Software Optimizations to Improve Performance

1.3.0 (2711) for OBi1 Series:
- Addresses a GV Backing Off Issue

1.3.0 (2690) for OBi1 Series:
- Improved audio-path synchronization on call establishment. Also fixes certain conditions where audio path is not initiated on incoming calls using the Google Voice communications service.
- Fixed: If the Caller ID coming from PSTN has a dash '-' inserted between the number(s), the Caller ID cannot be displayed on the telephone connected to the PHONE port.

1.3.0 (2669/2675) for OBi1 Series:
- Fixed Message Waiting indicator (Follow-on corner case fix in 2675)
- Polarity Reversal not enabled by default

1.3.0 (2651) for OBi1 Series:
- Various audio quality enhancements.
- Broadsoft interop enhancements.
- GV with SIP2SIS (PC-based SIP to Skype transcoding) re-packetization modification
- Added Parameter Option: "Assume connect after a short delay" to the LINE Port parameter: DetectOutboundConnectMethod
   Note: This setting invokes a 2 second connection delay when performing an AA callback via the PSTN line.

1.3.0 (2586) for OBi1 Series:
- Fixed AA callback issue where the caller, after connecting to the AA, then pressing one, cannot ring the phone.

1.3.0 (2575) for OBi1 Series:
Version 1.3 highlights:
- Paging Mode Support
- Star Codes for Blind Transfer (*98) and Barge-In (*96)
- Record up to 8 different customized AA prompts from the OBi PHONE Port.
- Upload / download of a package of user-recorded AA prompts via the OBi web browser interface.
- Call recording from the Call Status web page (PHONE port calls only):
   - Record and Stop Record button will be shown next to the call (Firefox Only).
   - Recording saved as an au file.
- Allow Caller-id spoofing for calls bridged via OBiTALK service. But use the obi number for circle-of-trust authentication.
- Support for blind transfer via star-code. Default star-code is *98
- All InboundCallRoute and OutboundCallRoute syntax will take a ";d=[delay-in-seconds]" parameter after the number to call from the specified trunk, and to insert a delay before the trunk makes the call. For example, SP1(18002211212;d=3) tells the OBi to call the number from SP1 after a 3-second delay.
- Support for options to insert Proxy-Require and Accept-Language headers in outbound INVITE, SUBSCRIBE, and REGISTER requests. Contents are configurable under ITSP Profile A/B - SIP section.
- Support configurable value of Max-Forwards header; option under ITSP Profile A/B - SIP section (Default: 70).
- Support for SIP over TCP/TLS.
- SRTP (S-Descriptor Authentication)

List of the 8 prompts that are replaceable with a user derived recording:
  - Welcome (to OBi Attendant) (optional)
  - Main Menu (optional)
  - Selection Menu – Press 1 to …, Press 2 to …
  - Enter PIN (optional if PIN not used)
  - Invalid PIN (optional if PIN not used)
  - Enter Number (followed by the # key)
  - Please Wait (while your call is being connected)
  - Goodbye (optional)
   - Other Post Dialing Announcements, following SIT tone, are not replaceable.
  Prompts can be recorded using the handset attached to the OBi PHONE port.
  Prompts will take effect after a reboot (automatically when hang up).
  The OBi web page has a buttons for the backup and restoration the AA prompts.

1.3 FAQ:
Q. How do I to record user prompts on my OBi device?
A. There are 10 slots for User Recordable Prompts
-   The maximum length for each user prompt is 60s
-   Total space available for all user prompts is 122s
-   User Prompts are referenced in AA Prompts configuration with the notations %USER1%, %USER2%, … %USER10%
-   Prompts are recorded via PHONE Port IVR main menu 0 (that is, dial * * * 0), and enter option 1001#
        for %USER1% prompt, 1002# for %USER2% prompt, and so on, up to 1010#.

When you are prompted to enter “value” when recording prompt, press any digit 0-9 to start recording. When you’re done, press #
o   Tips: Leave at least 1s of space at the end before hitting # to avoid the ending being truncated.
o   Afterwards, you can review your recording, and save it permanently if you are satisfied with it. Otherwise you can record again
o   After you have saved a recorded prompt, you can proceed to record a different one. When you hang up, the OBi device will reboot automatically so that the new prompts can take effect
-   Each user prompt has two status parameters shown at the top under the Auto Attendant page:
o   Description – A short text description of what to the prompt contains; limited to 80 characters. User can edit the description on the device web page. The description is stored as part of the prompt data and can be backed up and restored along with the prompt data. After you have recorded one more new prompts, you should go to this page to modify the descriptions and save them with the new prompts
o   Length – The duration of the prompt in milliseconds
o   The page also shows (at the bottom of the section) how much total space has been used and total available space remaining

-------

Q.  How do I use the recorded prompts for my OBi device's AA1 - Auto Attendant 1?
A.  AA1 has the following programmable prompt parameters with default factory provided contents:
o   Welcome – Default = “Welcome to OBi Attendant”
o   MenuTitle –  Default = “Main Menu”
o   Menu – Default = “Press 1 to continue this call, Press 2 to make a new call, Press 3 to enter a callback number”, repeated 3 times
o   EnterNumber – Default = “Enter number followed by # key”
o   EnterPin – Default = “Enter PIN”
o   InvalidPin – Default = “Invalid PIN”
o   PleaseWait – Default = “Please wait while your call is being connected”
o   Bye – Default = “Thank you for choosing OBiHAI. Goodbye”

-   The default contents are played if the prompt parameter is blank
-   To play your own recorded prompts, you enter one or more user prompts (separated by comma), such as: %USER1%, %USER2%. OBi will play the user prompts one by one in the order the user prompts are listed in the parameter
-   Each user prompt in the list takes an optional “r=[start][-end]” parameter to specify the range of the recorded prompt to play; unit is in milliseconds. For example:
o   %USER3%;r=1000      (starts playing at the 1000 ms mark to the end)
o   %USER5%;r=-2500      (starts from beginning to the 2500 ms mark)
o   %USER6%;r=1300-3720   (starts from the  1300 ms mark to the 3720 ms mark)
o   %USER5%;r=3200-1200   (does not play anything since end < start)
o  
-   To insert additional silence period while playing the prompts, you can add one more “&pause(<length>)” where length is the duration of the pause in seconds. For example:
o   %USER1%;r=1200,&pause(2),%USER2%,&pause(3)

-------

Q.  How do I share my AA prompts with others?
A.  You can backup your recorded prompts as a single file and restore it on any other OBi running version 1.3 or later release:
-   To backup the recorded prompts from the device web page:
o   Device Web page – System Management – Device Update - Backup AA User Prompts
o   Click Backup to save to your PC
-   To restore the saved prompt file back to an OBi (v1.3),  do it the same way as if you are upgrading the OBi firmware from the device browser and provide the path to the prompt backup file. The OBi can tell from the file header that you are trying to update the user prompts. WARNING: All the existing user prompts will be overwritten by the restored file, even if it does not include all the prompts. There is no “merge” function at the moment when restoring a prompt file.

Enhancements & Fixes in Maintenance Release 1.2.1 (2384):
- Touch-tone entry improvement when accessing a conference call bridge, entering credit card numbers, etc.
- Disallow *740x and *750x to write/read speed-dials (i.e., speed dial codes should not start with a 0).
- Added *76 to Clear a Speed Dial.
- Added ‘Outside Dial Tone’ to Tone Profile.
- Show reason code and phrase for unsuccessful outbound calls on SP1/2 and OBiTALK services in call history.
- Show Registration failure phrase (in addition to SIP RFC code) on the status web page.
- SkipCallScreening digit fix for Google Voice.
- Various other bug fixes.

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.

Note: The previous version of this information referenced a tool for uploading prerecorded audio prompts.  This tool is not available.  The only way to record audio prompts to the OBi is via the PHONE port.
« Last Edit: November 25, 2014, 09:55:59 am by MarkObihai » Logged

Obihai Technology (London, United Kingdom)
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