1/31/2024 0 Comments X lite softphone androidYou will need to make some associations between the two and perform some other ancillary activities in preparation. In CUCM you will need to create a SIP device and a user object. I like to get the station configured in CUCM before I start playing around with the client. The install is pretty straight forward and relatively quick. I suspect that procedures for 6.0 to 7.1 would be near identical.įirst, you will need to download the X-Lite application here. From looking at some of the examples on line (dating back to CUCM 5.0), the basic configuration we are going to discuss is pretty applicable to all CUCM appliance models. I like X-Lite, so I am using it as the example.įor my testing I used CUCM 7.0-1 (that is 7.1(3b)SU1 for the uninitiated). But, I hope to provide some examples to give you a better feel for how to add 3rd Party SIP endpoints to the CUCM. Every one of us has a slightly different environment that we operate in. I ask you, what am I supposed to do with that if I try to deploy this application in the real world? Now, I am not saying that I am going to cover all bases adequately in this example. The examples would have things like this: Second, the resources I have found on-line, including CouterPath’s own write up (which is a bit dated) and an article on Cisco’s “community” collaboration site, were just too light on details. X-Lite seemed like a reasonable choice as it was free and apparently pretty popular. I would definitely recommend using a prepackaged Asterisk distribution if you're new to VoIP stuff - AsteriskNOW should be fine, but I personally use the FreePBX distro in production and have had excellent results with it (and it's good at auto-configuring for NAT-related issues which saves me some headaches.First, I just wanted to add some 3rd party SIP phones to CUCM to start testing feature behaviors. In terms of Asterisk tutorials, these abound on the net (and Server Fault is not really a resource for "Find me a tutorial" - Google stays up to date on this stuff better than we ever could). Then slowly work your way up to dealing with traversing firewalls and NAT. So basically - start by getting things working inside your firewall. If you watch the data going back and forth you'll probably come up with a reasonable idea of where the problem is. NAT-related issues usually affect calls more than registration, and there is lots of info on how to deal with it if you google around.Ī good packet sniffer is your friend when troubleshooting VoIP SIP was never really designed to work with network address translation.Īgain, try getting things working on the same subnet, with no NAT or routing involved, before moving on to more difficult tasks. SIP (the VoIP protocol behind Asterisk) HATES NAT I would strongly advise troubleshooting on dedicated physical hardware. I don't know if they have specific guidance on this, but my experience is even in high-end VMWare environments you can get into all sorts of odd trouble with virtualized VoIP servers. Do you have any suggestions what we could do? Would may be wireshark be of any help? Any information, help or suggestion is much appreciated (also if you know good tutorials how to set up asterisk with freepbx)!Īsterisk does very poorly in virtual machines. Now, I know it's a very vague description of our problem and the error could be anywhere - but we don't even know where to start to look for the error. We configured the IP of the asterisk server as static. On the modem we forwarded all incoming requests on port 5060 to the local router and this router forwards all requests to the local IP of our asterisk-server. In X-Lite we specified the IP of the modem of the local network in which the asterisk server resides. But now, we would like to be able to access the server also from the internet - this is what we're strugglin with. We tried the configuration by registering some SIP-Phones via X-Lite in the local network, and it worked. Thus we have installed asteriskNow in a virtual environment (with virtualBox) on ubuntu 12.10. We currently have a school-project going in which we need to set up an PBX with Asterisk.
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