OK. I have a Polycom Soundpoint 650 and I recently was able to upgrade the software to 4.0.3 and I'm extremly happy with it. My only issue at the momment is being able to make mutiple calls per single line. Prior to upgrading the software I had my phone paired to our Switcvox system and it was running software 3.2.4. My current and previous SIP config is I have single extention on line 1 registered my company's pbx and then I've purchased my own private VoIP subscription and added it to line 2. I've created my own configurations for my phone and uploaded via TFTP however, I've used the web gui to configure the SIP lines. My issue that I'm having and this maybe a very simple fix is that when I'm on line 1 (company pbx) I use to be able to place a call on hold and if I pressed either new call or dialed the ext or number it would start dialing on line 1. I can recieve multiple calls on line 1 but cannot dial out if I am on a call or call is on hold. This is also true for line 2, both sip registrations are capiable of handling multiple outbound and inbound calls.
What am I doing wrong?
Solved! Go to Solution.
Hello jwilliams,
welcome to the Polycom Community.
I am not sure what your previous software was but try this parameter:
call.stickyAutoLineSeize="1"
A quick test in my lab shows this above parameter to bring back the "old" behavior.
Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.
Best Regards
Steffen Baier
Polycom Global Services
If official support is required please check how to phone or open a case here
----------------Hello jwilliams,
welcome to the Polycom Community.
I am not sure what your previous software was but try this parameter:
call.stickyAutoLineSeize="1"
A quick test in my lab shows this above parameter to bring back the "old" behavior.
Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.
Best Regards
Steffen Baier
Polycom Global Services
If official support is required please check how to phone or open a case here
----------------Thank you so much!
You are a God send and that was the fix. I have to re-create the sip.cfg file and once I did it resolved everything. I knew when I couldn't find the resolution that it would be a quick fix to resolve it.
About to work on a multi phone max configuration deployment setup so I'm sure I will be back but this is the reason why I love forums.
Thank you.
Just as a follow up,
have a look at this document => here <= which explains this feature in more detail.
Best Regards
Steffen Baier
If official support is required please check how to phone or open a case here
----------------