Taki 1.1 – multiple SIP accounts support

I’m pleased to announce new Taki release 1.1 which brings couple on new exciting  features and improvements. Major addition is support of multiple SIP accounts. Now you can consolidate all your SIP accounts and make/receive calls from one application. Another exciting improvement is enhanced NAT traversal support which should allow to overcome one-way audio issues.  Thank you to everyone who provided feedback and was diligent to report found issues. Please feel free to submit bug reports, feature requests or email me about found issues. Thanks!

Changelog for 1.1.0:

– Ability to add multiple SIP accounts (Lines) and the ability to select which line to call from. YES!

– transport: UDP/TCP.

– possibility to bind to any port (5060 by default).

– Possibility to enable/disable specific Codecs and set their priority.

– Advanced STUN, NAT traversal adjustments.

– Outbound proxy support.

– Adjust registrar and related parameters.

– Initial presence support.

– Improved UI responsiveness and performance.

– Help tab for easy access to Taki resources.

– SIP stack does not send 180 ringing so the caller does not hear ringing. FIXED

– If you hang up before “ok/ack” the user agent cancels the call but the GUI still plays the  ringing sound. You have to close the app and open it again to recover. FIXED

– Reject sends the user to voicemail but keeps ringing have to close the application to clear it. FIXED

– When the keyboard on the playbook is in view the numbers and letters on the dial pad are not clearly visible. This is only a problem in landscape mode. FIXED

– Call history->cdr->name is not populated in outbound calls. FIXED

– ignore does not work(not sure what it should do). FIXED (stops local ringing without rejecting the call)

– many other fixes…

8 Comments

  1. Help,
    Just purchased but wont make a call for me….tried skype, ekiga and couple other sips.

    Tried to dial my cel # but does not work.

    My skype program on desktop does dial the cel # fine.

    • What SIP provider are you using?
      Were you able to configure your account and make it registered?

      • Hi,

        I need help to configure my Taki. I’m not familiar with this. I would like to have an exemple to fill right the register section. Thanks.

        • In most cases it’ll be enough to fill parameters that you see on Settings tab. Please ask your SIP provider for this information.
          If you need to specify Registrar server which is not the same as Domain name you should be able to do it at Account Advance Settings, but that’s a rare case.

  2. I set up a skype sip profile and i configured a profile taki and registered it. I see the registration in my skype manager.

    When i dial a number iimmediately get a message “call ended: connection reset by peer”

    Any suggestions?

    Thx,
    Gary

    • Please enable SIP messages and send me your call logs.
      Try dialing any USA Toll Free number.

  3. Hi,

    I configured sip on my playbook but could not make a call. On the screen it shows registered. The program freezes. I have to restart my playbook to get going. Is there any problem.

    The sip provider: sip.voipgo.com

    Thanks

    • In “General SIP Settings”-> “Media” please try to leave active G711-ulaw and Speex-nb codecs only. Rest can be disabled.
      After saving settings please try to make a test call.

      If you still have issues after that you can send me a logs with enabled “SIP messages” from a Log tab.