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SIP Phone Plugin slow to answer

So, I have the SIP phone plugin working with Asterisk 1.4 (Openfire 3.3.2, Sip 1.0.2, Spark 2.5.8) except that if I pick up a dialed phone it takes Spark about 10 seconds to notice, stop ringing, and patch through the audio. Once that happens everything works well except the presence status stays at “On Phone” so that I have to manually change it back to available. Anyway, this plugin is very close to being useful and if anyone can solve those two issues (mainly the delay since presence for us isn’t as critical) that would be great .

BTW, the Asterisk server is on the same LAN with no firewalls, etc, in the way.

OK, I just realized the problem with Presence and that was from testing the Asterisk-IM plugin with the same extension thus creating conflicts. I removed that and the Presence worked fine. So, just the delay is left as a problem.


Check this Thread: http://www.igniterealtime.org/community/message/142631#142631



thanks for the link which gives a work-around (disabling NetBIOS over TCP\IP), it works like a charm. Of course, I’d prefer to not have to use that work-around as we still need NetBIOS in certain circumstances for legacy machines. My other SIP clients (X-Lite, SJPhone, etc) don’t seem to need this workaround. So, hopefully the next iteration of this plugin will look into the issue.

Thanks though for the help,


This is an issue which must be corrected. We cant all go around and start disabling such options in network configurations on client machines. This isnt convenient.

Do licensed users have to deal with such workarounds? or does DEV correct it for them.

I ask because I will be a licensed client in about a week.



To eliminate this delay, we suposed to have access to JMF core which is closed ( NOT OPENSOURCE ).

The solution would be changing the media API, but change to which one?

There is some Java APIs but mostly incomplete APIs. We still searching for fixes on this without changing the code of JMF and without end user’s interation like “Disabling NetBios”.

Best Regards,


Had the same Problem with the Trixbox set up in a DMZ.

Created a new zone on my DNS server for the DMZ subnet, then created the Reverse PRT record for the trixbox.

As soon as that was created, the delay went away both on diailing out and on actually being able to speak. (Netbios is still running).

Can someone else try this and confirm if that works for them too?


it seems SIP Communicator algo uses JMF but don’t have this delay on sound startup. Anyone ever tried to compare both implementations to find out the difference and fix it ?