Access tio SIP and calling functions in 2.4.1+

I’'m playing with WF3.2RC2 and Spark 2.4.1 with SIP calling. First, very impressive all around- everything looks great!

Second, I’'m curious as to what access plugin developers have to the SIP stack and calling functions. For example, could I have an address book plugin with which I could kick off calls to the telephone numbers of my contacts by right-clicking and selecting a number? Likewise, is there any way to access call state so that I could, for example, take intelligent action if a second inbound call arrives?

Finally, are you planning on adding functionality such that I can right-click on IM contacts (buddy list) that have telephone numbers in their V-Cards and call them direct?

Once again, great stuff!

Thanks,

Dan

Hi ecodan,

Thanks for kind comments. As for your questions, we are planning on providing address book functionallity in the very near future post 1.0 release. As for the right-click functioanlity, Spark already takes a look at the users vCard for home/mobile/work numbers and will display all options depending on what has been filled out. Spark even uses the vCard information for transfering of calls.

Cheers,

Derek

we are planning on providing address book functionallity in the very near future post 1.0 release.

I just used AB as an example… I can think of quite a few apps that could benefit by having access to a SIP UA. Is (or will) the UA in Spark accessible to plugins for call initiation and/or call state awareness?

As for the right-click functioanlity, Spark already takes a look at the users vCard for home/mobile/work numbers and will display all options depending on what has been filled out. Spark even uses the vCard information for transfering of calls.

I figured this is how it should work, but it doesn’'t seem to be working properly. I have 2 users on the WF server, A and B. Both A and B have V-Cards configured through the Spark->Edit My Profile menu. Both have images uploaded and I see the images. I verified the data in the database. On both A and B, however, when I go back to Edit my Profile, the profile appears to be completely empty. If I enter new data, it wipes out the old vcard and replaces it with the new, but the data still does not show up to the other contact, and when I go to Edit My Profile it appears empty again.

On the SIP client, I played around with it some more, hitting various proxys and media servers in our lab. A couple of comments:

  • The presence change to “On Phone” doesn’'t see to kick in until several minutes into the call

  • The presence change back from “On Phone” once the call ends doesn’'t seem to ever happen

  • Every once in a while the softphone bar on the GUI disappears and I see “Registering” with the spinning arrows. When I go to Actions, I see that the “Phone Enabled” option has deselected itself. When I re-select it, the softphone re-appears.

  • The audio quality seems to be OK. but fairly jittery. I see that G.711 is the only codec presented… is there any config to increase the jitter buffer?

I’'m going to keep using it and will report anything else I find (unless you ask me to stop ).

Cheers,

Dan