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Okay, so I recently went live with a hosted VoIP solution. Works great, especially since they are polycom HD phones granted you only get HD quality to another HD phone user. Scenario, we have 3 locations, all using FG80c with with 4MR1 Patch 4, and each user has a VoIP phone using SIP traffic UDP 5060. Session-Helper has been deleted and ALG has been applied to outgoing SIP traffic on our voice vlan.
Anyways, there are two options to managing the phone via the computer. A web based call manager gui that connects to the hosting servers or a toolbar add-in to either IE or Mozilla, or MS outlook. 1 problem exist at each location, if user 1 in location A calls user 2 in location A via toolbar, both parties pick up the phone, there is no audio. However, if either user puts the call on hold, and then hits resume, the audio now functions.
Again, this only happens within the site when a user calls another user via the toolbar addin or the reception/operator software installed locally on the operator' s PC. SITE to SITE works beautifully.
Any thoughts? I' m considering going to 4.2 patch 2, but i wanted to see if anyone has seen this. My provider just tells me it' s a firewall issue and I' m trying to get in touch with someone there who can provide a little more insight.
Per Host -The toolbar simply contacts the hosting server to call originators phone, then call user b, if call is internal, leave sip traffic inside the site.