[Asterisk-Users] Re: Polycom IP 300/500 Conferencing Behavior

[Greg Boehnlein]

On Fri, 21 Jan 2005, Greg Boehnlein wrote:

> Hello,
> 	I've got a mixture of SPIP 300 and 500 phones in production for 
> various clients. I've got the XML settings configured for local 
> conferencing, but I'm not seeing the expected behavior from the phone when 
> I attempt to conference two calls together. According to the manual, while 
> talking to the first party, you simply hit Conference, dial the second 
> party and then Conference to join them. This is supposed to put the first 
> party on Hold until you bridge them together with the second press of the 
> Conference button.
> 	That is all fine and well, but it doesn't quite work the way that 
> the manual describes. Instead of joining the two calls together when the 
> Conference key is pressed for the second time, the first party is taken 
> off hold and hears dead silence. The only way to correctly join the 
> parties is to hit the Hold and then Resume soft key, at which point all 
> three parties can talk to each other.
> 
> As an illustration
> 
> Conf -> Dial -> Conf doesn't work.
> 
> However,
> 
> Conf -> Dial -> Conf -> Hold -> Resume DOES work.
> 
> 	I'm running 1.3.4 firmware on all the phones, and I can't for the 
> life of me figure out what is causing this problem. It is very likely some 
> misconfiguration in the XML files, but I can't find it. Anyone have any 
> suggestions?

Hello,
	I just thought I would follow-up on this post and mention that 
somwhere between Jan 21st and today, the Conferencing issue that I 
described below is no longer an issue. Normally, I wouldn't include the 
entire quoted context of the message for bandwidth reasons, but in this 
case, since the topic is nearly 2 months old, I figured it would be 
helpful to keep things consistent and on-thread.

	I'm not sure what may have been fixed. I.E. I don't know if it was 
a patch to chan_sip in stable or what. All I can tell you is that I 
haven't made a single change to either sip.conf or my XML config files, 
since the original posting in January. However, I have updated asterisk 
several times from the 1.0 branch.

I'm happy that this is fixed, but I am going to do a little more reasearch 
to see if I can get it to fail again by backdating chan_sip and 
incrementing it forward.

This should be interesting. :)

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