I am using soundpoint IP 331 phones running version 3.3.1.0769. We have two asterisk servers (13.2). Each phone is configured to have a registration to each server. This maps to 2 sip servers with a single reg in the config files. We are doing some testing on our primary server, so it is up and available, however I have my numbers pointed at our backup server. I set the outbound proxy for line 2 on my phones to the backup server so that when placing an outbound call I can force the call to the backup server. The only issue I have not been able to resolve is with transfers. The issue is as follows:
A sip call from "A" comes into the backup server via one of our SIP providers. This call is connected to "B" in our call center. "B" decides to escalate the issue and transfer to manager "C".
If "B" initiates a blind transfer to "C", the phone will do so via the same registration the initial call came on it (the backup server).
If, however, "B" decides to do an attended transfer, the phone will place the call out of what appears to be the first available line which ends up sending the call to the primary server. "B" is able to have a conversation with "C", but when they attempt to complate the transfer, it fails. I know that I can also configure remote transfers on my asterisk servers, however I would prefer to lock transfers to a single server if possible. Is there a way to configure the phone to place this second call using the same registration, forcing it to the same server?
Thanks for any advice anyone can provide.
I have a bit more information to add. After more reading, I realized that I don't likely want to use the failover functionality as I am not using redundant SBC in front of a sip application server, I have two separate sip application servers that I can switch between. To resolve that, I now use the reg.x.server.address= to create 2 registrations, each one tied to a single server. On the phone I can see that line 1 is registered to server a, and line 2 is registered to server b in the settings, however when it receives a call from server b it shows on line 1.
I then went ahead and updated the user on server b to make it different (added a -1 and -2 to the users on each server) and tried again. Now, a call from server a shows on line 1, a call from server b shows on line 2, and transfer works correctly! It looks like even though I'm trying to explicitly say they should be two separate registrations calls from server b show on line 1 if the username is identical.
I noticed the following correction in the release notes for 3.3.2. Can anyone confirm whether this correction is meant to address this issue?
67867: The phone seizes a wrong line after transferring an incoming call to the line when going off-hock.
I'm new to the polycom world, so please forgive me if any of these questions seem basic. If I've missed something in the admin guides or documentation I appreciate even a point in the right direction. Thanks again!
welcome to the Polycom Community.
Firstly, UC Software 3.3.1 or 3.3.2 is not a currently supported version.
The latest 3.3.x is 3.3.5 but from a support perspective the SPIP331 should be running 4.0.8
The community's VoIP FAQ contains this post here:
Oct 7, 2011 Question: How can I setup my Phone / Provisioning / Download / Upgrade / Update / Downgrade Software?
Resolution: Please check => here <=
If you want a redundant setup then I suggest to create a DNS entry for both servers and use a FQDN as the main server for the Line in question. If that server goes down the DNS will resolve the IP of the other server.
Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.
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