Hello I've been using PLANET IP Phones (PLANET VIP-254PT) in my office and at home. I've been using basically only "the IP" for VoIP connection over the LAN (never used or subscribed to any VoIP service provider or something similar, because there's no need) I use Dynamic DNS service and STUN Server for VoIP over Internet (again, no need to use any VoIP service provider or something similar, because there's no need) and assign several different port numbers (like 5060, 5061...) if I'd like to make several IP Phones inside the LAN reachable from the Internet. So far so good. But lately a need has arisen to forward the calls coming to my PSTN line to these IP Phones. To set it up, I bought 1 second hand (unlocked) Linksys SPA 3102 and 1 SPA 2102 . After tinkering with them for several hours I noticed that their settings are more complex than PLANET's, especially the thing called "dial plan" Now before I 'm able to make a PSTN to Ethernet connection, I'd like to make a simple VoIP call from SPA 2102 to one my VIP-254PT phones, by directly dialling the IP # of VIP-254PT or vice versa but this cannot be done! Both phones are on the same LAN. SPA2102 ATA IP is: 192.168.1.202 SIP Port #: 5060 Router >> Internet Connection Settings Connection Type: Static IP: 192.168.1.202 & LAN Networking Service: Bridge. Voice >> Line 1 >> Make call without reg: yes Ans w.o. reg: yes, Enable Ip Dialling: yes, User ID: SPA2102 Dial Plan : I have left this section empty as I don't know the details. VIP-254PT IP is: 192.168.1.221 SIP Port #: 5060 Under this setup when I dial 192*168*1*202# the VIP-254PT display says "unknown number" So something's missing on SPA 2102 most probably the Dial Plan string. I think I have to form my own string there. How to form it? I'd like to make it such that SPA2102 will allow every IP Phone on the LAN (and preferably on the Internet, too) Thanks.
You can start with a dial plan like (<#1:>S0<:192.168.1.212:5060>) Where you dial #1 to call 192.168.1.212:5060. The S0 means set the short timer to zero and dial immediately. You don't need the 5060 because that is the default but it doesn't hurt to put it there. You do need the port number if it isn't the default 5060. The #1 before the colon is stripped before dialing. If 192.168.1.212 has a userid configured on the device you need to include the userid: userid@192.168.1.212:5060. The other thing you can do is setup speed dials. You can have up to 8 speed dials that you setup on the User Tab. You dial a speed dial by dialing the speed dial number (1 digit) plus the # sign. A 1 character element in the dial plan will allow speed dials (x|<#1:>S0<:192.168.1.212:5060>) You can put in an element for each phone if you wish separated by | character. I think there is a way to dial the ip address itself however I haven't tried that. You don't have to but I would setup everything with a different port number. The number itself isn't significant. You can use most anything. The ATA Administration Guide has some more detail on dial plans as does the "Sticky" thread in this forum about dial plans. http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf
Hello Thank you for the comprehensive comment. I've put the (<#1:>S0<:192.168.1.221:5060>) in the dial plan section and it works; #1 calls the 192.168.1.221, which is the PLANET IP Phone. That's fine. But there are still two things. 1- I still cannot call the SPA2102 (IP: 192.168.1.202) back from PLANET IP Phone. The standard procedure I use is to directly enter the IP of SPA2102 using the numpad of PLANET IP Phone like 192*168*1*202#, where * 's are interpreted as . and # as "send." When I dial the SPA2102 in this way from PLANET IP Phone, its display says "unknown number" 2- I cannot call the "Line 1" of SPA3102 (IP: 192.168.1.32) either from SPA2102 or from PLANET IP Phone. An analog telephone is connected to the RJ11 Phone jack of SPA 3102. When I add (<#2:>S0<:192.168.1.32:5060>) in the dial plan section of SPA2102 ,so that the resulting sting is (<#1:>S0<:192.168.1.221:5060>|<#2:>S0<:192.168.1.32:5060>) and dial #2, the analog telephone connected to the SPA3102 doesn't ring. FYI, the analog telephone connected to the SPA3102 does ring when an PSTN call comes to SPA3102. Could you give some instructions on how to proceed? Thanks.
First make sure you can ping the ip address of the SPA's from a computer on the network. Insure there are no local network problems. You may need to forward the port numbers in your router for local network calls, you definetely do if you are going over the internet. This is because the calls are not solicited, there is no registration. Make sure there is no UserID setup on the SPA's. If there is a UserID setup it must be included in the call address. Of course you need Ans Call Without Reg: Yes setup. Then I would start using the sip debug trace function of the SPA's. You need to look at the Sip Invite and see if it looks OK and then you need to see if there is a response, if any, to the invite from the called adapter. Get the SPA2102 to SPA3102 working first because you can run traces on each on separately to see what is happening. If you need a reference on what a Sip Invite looks like you can look at your example that works. To run the sip debug trace you download and install a Syslog program on a local computer. Then you put the computer's ip address on the SPA's system Tab under Debug Server and you set the Debug Level to 3. On the Line Tabs and the PSTN Line Tab you set the Sip Debug Option to FULL. The syslog program saves a file to the computer's hard drive. You can download a simple syslog program for a Windows PC here: https://supportforums.cisco.com/docs/DOC-9862 Edit: If you can dial different port numbers on your Planet IP Phone, you might consider setting different port numbers on the SPA3102 and SPA2102 just in case your router is getting confused. You might also change the RTP Port Ranges in the Sip Tab also. You can use any numbers or number ranges, just be sure to use them in the addressing.
Hello and thank you for the help. Now I have resetted and rearranged the settings of SPA3102 and I have managed to call the "Line 1" of SPA3102 (IP: 192.168.1.32) from SPA210. That's fine. But I still cannot call the "Line 1" of SPA3102 (IP: 192.168.1.32) from PLANET IP Phone, though I have set "Ans Call Without Reg" of Line 1 as yes. In this case I might try debug trace using the way you mentioned but for now I must postpone that as it 's likely to take some time. Note on your "Edit: If you can dial different port numbers on your Planet IP Phone, you might consider setting different port numbers on the SPA3102 and SPA2102 just in case your router is getting confused." I don't think I need to set different port numbers for SPA3102 and/or SPA2102 because all of the IP phones I talked about so far are on the same LAN, ie they don't pass thru any router. -------------------------------------------------- I had one side effect after I re-edited the SPA3102: It (the analog phone connected to Line 1) no longer takes the dial tone of the PSTN line. I must be so because I have set Make Call Without Reg: of Line 1 as yes and Dial Plan as (<#1:>S0<:192.168.1.221:5060>|<#2:>S0<:192.168.1.202:5060>|<#3:>S0<:192.168.1.211:5061>) to be able to call all those 3 IP Phones easily and in this Dial Plan, I can really call them easily but this time as I've said, it no longer gets the PSTN dial tone and I cannot call a PSTN number. Hoever, it accepts PSTN calls perfectly. The initial (factory-set) Dial Plan was: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) To make SPA3102 take the PSTN dial tone, I have to set the "Make Call Without Reg: of Line 1" as no but this time I cannot dial the IP Phones using the #1, #2, etc. So the question is, how to make both sets of phones (the IP Phones on the LAN and other far away phones on the PSTN) phones diallable from the Line 1 of SPA3102?
If you can call the SPA3102 from the SPA2102 then I think I would need to see the trace from the ip phone to see what is wrong with calls coming from the ip phone. The trace, however, might not show anything if the SPA3102 rejects the incoming call and that might present a debugging problem unless the Planet phone has some debugging tools like a syslog or unless you can setup a packet sniffer like WireShark on the network that shows the sip signalling. If you have no router to issue DHCP ip addresses then you only have a switch? Is the SPA3102 setup with a static ip address? Why don't you post your SPA3102 configuration. With your computer's web browser you save it to your computer's hard drive (single save) as a .htm, .html or .mht file. Make a .zip file of it and post it to a posting. You have to add an element to the dial plan to access the PSTN line from Line 1. Something like this: ((<#1:>S0<:192.168.1.221:5060>|<#2:>S0<:192.168.1.202:5060>|<#3:>S0<:192.168.1.211:5061>|xx.<: @gw0>) which will send any call to the FXO port unless you dial #1, #2, or #3 initially. You don't get the actual dial tone from the PSTN line, you get the dial tone from the SPA3102 and it sends the call to the FXO port. I would have Register: No in addition to the Make and Ans Call Without Reg: Yes settings. I guess when you have Make Call Without Reg: Yes the SPA3102 throws it's no power relay to connect you directly to the PSTN Line.
