Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list. First, a little on what I'm trying to do: I need to setup the PBX to answer on multiple 'lines' (I use that word with trepidation as I'm not sure if it's the right term in the absence of modems & actual lines) and play a brief message identifying itself as the 'line' connected to. The originator of that call will be a softphone. Before rolling this out to my lab, I'm trying to work out the proper config on my laptop. Therein I have Windows XP w/ VMware - Red Hat 7.3 is running in a VMware session. My connection to the net is NAT'd. My internal IPs for both XP & Linux are of the 192.168.x.x private network variety. I have the Xten X-Lite softphone on XP to test with. (I also have another called SJPhone, but haven't done much with that past installing it.) I've configured a number for that through freeworlddialup.com. X-Lite appears to be working fine. At least I can dial their echo & test numbers without a problem and get the expected responses. So the questions: 1. A general "will this work?" (vmware linux, same pc as phone, NAT'd addresses,etc) 2. Has anyone done it this way before and/or followed Andy Powell's doc, and have any suggestions or things to watch out for? 3. Reading the various published SIP documentation (Ubiquity's 'Understanding SIP' for instance), it seems like freeworlddialup is acting as Registrar, Proxy & Redirect server. Is that accurate? 4. How do I tell the freeworlddialup registrar 'where' to find my PBX? Should I setup an account from it - like I did with the softphone on XP - so it will have a 'phone number' of its own? Or is the proxy/redirect server expecting to talk to the Asterisk PBX in some other way? I appreciate any and all responses. Please cc my email address directly on replies as I have the list configured in digest mode to stem the flow a bit and don't want to miss any of them in the mix. Thanks to one and all Fletcher Bonds Operations Software Tester TeleCommunication Systems, Inc. (TCS) Enabling Convergent Technologies www.telecomsys.com Fbonds@seattle.telecomsys.com
for question 4. You need to register with fwd first, then use registry => command in sip.conf. aaron On Fri, 25 Jun 2004 08:32:07 -0700, Fletcher Bonds <fbonds@seattle.telecomsys.com> wrote:> > Good morning all, > > I'm setting up Asterisk for the first time with no prior PBX experience. > I'm following Andy Powell's 'Getting Started with Asterisk' > (http://www.automated.it/guidetoasterisk.htm). This is my second time > through that document - as I did something weird the first time and really > upset it somehow - and I wanted to ask a few general questions of the list. > > First, a little on what I'm trying to do: I need to setup the PBX to answer > on multiple 'lines' (I use that word with trepidation as I'm not sure if > it's the right term in the absence of modems & actual lines) and play a > brief message identifying itself as the 'line' connected to. The originator > of that call will be a softphone. > > Before rolling this out to my lab, I'm trying to work out the proper config > on my laptop. Therein I have Windows XP w/ VMware - Red Hat 7.3 is running > in a VMware session. My connection to the net is NAT'd. My internal IPs > for both XP & Linux are of the 192.168.x.x private network variety. I have > the Xten X-Lite softphone on XP to test with. (I also have another called > SJPhone, but haven't done much with that past installing it.) I've > configured a number for that through freeworlddialup.com. X-Lite appears to > be working fine. At least I can dial their echo & test numbers without a > problem and get the expected responses. > > So the questions: > > 1. A general "will this work?" (vmware linux, same pc as phone, NAT'd > addresses,etc) > > 2. Has anyone done it this way before and/or followed Andy Powell's doc, and > have any suggestions or things to watch out for? > > 3. Reading the various published SIP documentation (Ubiquity's > 'Understanding SIP' for instance), it seems like freeworlddialup is acting > as Registrar, Proxy & Redirect server. Is that accurate? > > 4. How do I tell the freeworlddialup registrar 'where' to find my PBX? > Should I setup an account from it - like I did with the softphone on XP - so > it will have a 'phone number' of its own? Or is the proxy/redirect server > expecting to talk to the Asterisk PBX in some other way? > > I appreciate any and all responses. Please cc my email address directly on > replies as I have the list configured in digest mode to stem the flow a bit > and don't want to miss any of them in the mix. > > Thanks to one and all > > Fletcher Bonds > Operations Software Tester > TeleCommunication Systems, Inc. (TCS) > Enabling Convergent Technologies > www.telecomsys.com > Fbonds@seattle.telecomsys.com > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
On Fri, 25 Jun 2004, Fletcher Bonds wrote:> 1. A general "will this work?" (vmware linux, same pc as phone, NAT'd > addresses,etc)You'll probably be the first person to try it. I'd guess that it will work, but expect call quality to be impacted because of all the extra scheduling and virtualisation. You will need to make sure that UDP streams from your linux side can get through the Windows with NATTing, and importantly that replies get back. Port 5060 needs to be "permanently" directed through to Linux. You set another range of ports in rtp.conf (rtpstart and rtpend) - these will also need directing through to Linux.> 3. Reading the various published SIP documentation (Ubiquity's > 'Understanding SIP' for instance), it seems like freeworlddialup is acting > as Registrar, Proxy & Redirect server. Is that accurate?With SIP there are all these terms - but practically speaking you send SIP messages somewhere, and get replies. FWD is a place which you talk to using SIP. I guess they are a registry (you can send REGISTER packets). They aren't generally a proxy - usually your audio streams end up going directly to the other endpoint. They do have a proxy that you can use if you need to because of NAT. You might need it. And I don't know what a redirect server does.> 4. How do I tell the freeworlddialup registrar 'where' to find my PBX?You put a register line in your sip.conf> Should I setup an account from it - like I did with the softphone on XP - so > it will have a 'phone number' of its own?Yes - you can open another FWD account, or you can use the one you setup for your XP softphone.> Or is the proxy/redirect server > expecting to talk to the Asterisk PBX in some other way?FWD is happy to talk to you with SIP. They do also have a test connection using IAX, Asterisk's own protocol - but why not leave that for another day.> > I appreciate any and all responses. Please cc my email address directly on > replies as I have the list configured in digest mode to stem the flow a bit > and don't want to miss any of them in the mix. > > Thanks to one and all > > Fletcher Bonds > Operations Software Tester > TeleCommunication Systems, Inc. (TCS) > Enabling Convergent Technologies > www.telecomsys.com > Fbonds@seattle.telecomsys.com > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >