![]() ![]() The SBCs handle traffic between our network and the PSTN, not between us and your site. You do NOT have to list the IPs of the SBCs in your firewall. This works within the browser and should not require special configuration.Ĭomm Client works within the browser and communicates to a cluster via TCP and should not require special configuration. Presence information (red/green registration status indicators) and User Dashboard information is sent to your web browser from the presence server. A quick google search for SIP ALG detector will find you a couple web tests and. The PureEdge web portal server IP is 64.94.196.58. SIP ALG is something you dont want in a firewall or router. This way I could keep the other functioning and now I know where to look if I need to add or remove any specific one. I thought that was the best solution since SIP ALG is the problem, not all ALGs. Grandstream 2000 phones provision via TFTP to 64.94.196.99. Ive removed the SIP lines from the /etc/modules.d/nf-nathelper-extra Rebooted and ran the SIP ALG detector. These IPs can change at any time, and we recommend allowing via ports rather than IPs.ĭepending on the method of provisioning this will traverse on these ports:Ĭisco 7940/7960 phones now also provision via TFTP to 184.169.138.90. , which is load balanced between these 4 servers: The ports can be seen by going to the extension detail page and clicking Show Details.Īn exception to this is the Edgemarc router which will handle all traffic on 5060 WAN-side by default. * Most SIP equipment works on port 5060 natively, but your router will usually NAT most phones to other ports on the WAN side. SIP and RTP Server: Feature Server hostname is listed in Customer Settings page, Service tab.RTP Server: Feature Server hostname is listed in Customer Settings page, Service tab ping the hostname to get the IP.SIP Server: SIP Proxy server hostname is listed in Customer Settings page, Service tab, as the Registration Server ping the hostname to get the IP.If it is Access SBC or SIP Proxy, SIP will originate from SIP Proxy IP, and RTP from the Feature Server IP.If the type is Feature Server, both SIP and RTP will originate from that IP.Since the introduction of SIP Proxy these may originate from different IPs depending if the Registration Type is SIP Proxy or Feature Server. The most important services for VOIP are SIP (Registration and Call Control) and RTP (Audio). If you are occasionally having problems receiving calls, this is typically the culprit as the router is closing the connection before phones have a chance to re-register. If this setting exists, it should be at 300-seconds or longer. Many routers also have a UDP Timeout setting. You may download and run the SIP ALG Detector linked below to help in your discoveries. If phones are having problems registering, disable any feature that references SIP ALG, SIP Helper or most likely anything that references SIP. SIP ALG's are enabled by default on many router models and may also be enabled on the ISP's equipment. There can be no SIP ALG's enabled on any devices between the phones and the TelNexa cloud system. ![]()
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