![]() But when I look at the VOIP Server it doesn't recognize any phones outside of the 8.0/24 network. But the Server doesn't know how to get back to them which seems weird. ![]() they seem to be able to dial out appropriately. I made their journey 1 step closer by configuring the phones to look at the 8.212 voip server. The phones were on the 2.0 network at the old office, and are on the 30.0 at the new office. ![]() But the asterisk seems to not know how to establish what ip address the extension has. This might be the solution, and maybe i'm thinking wrong here. I'm really not sure what i'm missing in this puzzle. I imagine if i fixed it so people could call site D, people would then be able to hear audio. The problem seems to me not that they are on a different network or that they dont have name resolution it seems to me that other network's don't know how to get back to the site D through the bovpn to establish the full connection for the phone call. I do not have a local WINs/DNS server on the remote network? I am curious if this would solve my problem? Or possibly just set up DHCP relay through the bovpn to site A's dns server which seems to work just fine. I can ping or remote into the local phones on the 30.0 network So the BOVPN seems to be set up properly. This network is reachable by any device on the network by ip address, Site D is reachable by all other sites, and all other sites are reachable by site D via ip address pings. The phones can call out successfully, but when you pick it up their is no voice. Site D is the site with the BOVPN set up where the phones aren't working. Site B has a secondary asterisk phone server and currently has a leased line. Site A is our main site it has a phone server it also is housing the primary watchguard Various other trademarks are held by their respective owners.I just set up a bovpn this past week thu / Fri and can't get the phones to fully communicate. WatchGuard and the WatchGuard logo are registered trademarks or trademarks of WatchGuard Technologies in the United States and other countries. If you do not want to log connections made by a user with an access level exception, clear the Log check box adjacent to the exception. To delete an exception, select it in the list and click Remove.Ĭonnections made by users who have an access level exception are logged by default. These settings apply only to SIP VoIP traffic. You can select whether to allow users to Start calls only, Receive calls only, Start and receive calls, or give them No VoIP access. įrom the Access Level drop-down list, select an access level and click Add. This is usually a SIP address in the format such as. To create an exception to the default settings you specified, type the Address of Record (the address that shows up in the TO and FROM headers of the packet) for the exception. To create a log message for each SIP VoIP connection that is started or received, select the adjacent Log check box. To allow all VoIP users to receive calls by default, select the Receive VoIP calls check box. To allow all VoIP users to start calls by default, select the Start VoIP calls check box. When enabled, the SIP-ALG allows or restricts calls based on the options you set. ![]() To enable the access control feature, select this check box. SIP-ALG Action access control configuration in Policy Manager SIP-ALG Action access control configuration in Fireware Web UI ![]() In the SIP-ALG Action Access Control configuration, you can create a list of users who are allowed to send VoIP network traffic. ![]()
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