| |
| News & Reviews |
Welcome to the Voxilla VoIP Forum.
Voxilla has been a trusted source for accurate, up-to-date information on the IP Communications industry since 2002. A dedicated staff of reporters and engineers produce feature articles and product reviews to keep industry watchers abreast of the people, companies, and trends driving a fast moving market.
You are currently viewing our boards as a guest which gives you limited access to view most discussions and access our other features. By joining our free community you will have access to post topics, communicate privately with other members (PM), respond to polls, upload content and access many other special features. Registration is fast, simple and absolutely free so please, join our community today!
If you have any problems with the registration process or your account login, please contact contact us.
Voxilla VoIP Forum |
Configuring 2 SPA 3000 to "bridge" calls between DTechnical support, how-to guides, troubleshooting, and general assistance for Linksys hardware. |
| | LinkBack | Thread Tools | Rate Thread | Display Modes |
| |||
| I have 2 SPA 3000s. One (in DC) is connected via static public IP to the net and and to a standard POTS line the second (in Rome) as yet has no config and just a handset. I would like to set them up so that when I dial the DC POTS number it rings in Rome and when I pick up the phone connected to the SPA 3000 in Rome I actually have my DC phone line to make calls on. I was hoping that I would be able to use the wizard to achieve this but it looks as if that is not possible. Can someone point me in the right direction on how to achieve this. I am new to VOIP but consider myself technically compteant otherwise. Thanks in advance |
| |||
| Thanks for the pointers I have played with this all day to no great avail. I have attempted to follow your pointers but have more questions than answers to this point. (I take it that DC SPA3000 PSTN port equates to the PSTN Line tab) I have enetered the following on DC SPA PSTN Line basic tab Line enable = yes SIP Port = 5061 Proxy and Registration Proxy = blank Register = yes Make call w/o Reg = no Ans call w/o reg = no Subscriber Info Display name = Roma User ID = Ken Password = something-very-secret Use Auth ID = yes Auth ID = Ken SWITCHING TO ADVANCED VIEW I ADDED Dial Plan 6: = SO<:Ken@IPRomaVOIP:5060> VoIP-To-PSTN Gateway Setup VoIP-To-PSTN Gateway Enable = yes VoIP Caller Auth Method = HTTP Digest Line 1 Voip Caller DP = 6 Default VoIP Caller DP = 6 Line 1 Fallback = 6 VoIP Users and Passwords (HTTP Auth) VoIP User 1 Auth ID: = Ken VoIP User 1 DP = 6 VoIP User 1 Password = Something-very-secret PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable = yes PSTN Caller AUth Method = none PSTN Ring Thru Line 1 = yes PSTN PIN max retry = 3 PSTN CID for VoIP CID = no PSTN Caller default DP = 6 I have a static public IP with all DNS and Gateway info working For the Rome SPA (I actually have the machine here I am trying to preconfig it and will then send it) I tried it with both DNS and static public IPs This is what I entered On the basic Line 1 tab I enetered Line Enable = yes SIP port = 5060 Proxy = (the DC SPA static IP) Register = no Make call w/o reg = yes Ans call w/o reg = yes Subscriber Info Display Name = Roma User ID = Ken Password = something-very-secret Use Auth ID = yes Auth ID = Ken SWITCHING TO ADVANCED VIEW I ADDED use outbaound proxy = yes outbound proxy = (the DC SPA static IP) use DNS srv = yes When I call the DC POTS line I get a ringing tone and then a busy signal If I use the DC POTS line to place a call I get a quick busy signal The DC SPA reports Last Called VoIP Number = 127.0.0.1:5060 Last PSTN Disconnect Reason Disconect Tone Any help would be much appreciated!! macudc |
| |||
| Hi, Just another user with a similar usage and similar problem, just wondering if you made any progress. I have played with these settings untill I was blue in the face. (really I am just pale and need to get out more) It does seem like I am missing something, any other thoughts. thanks in advance |
| |||
| NO progress. I am also blue in the face. I have tried everything I can think of, including giving both machine static IP (to take any firwall problems out of the mix) with no joy. I will shortly have to return these units, unless I can get help from somewhere else. I will post here if I find anything. Let me know if you get anywhere... what kills me is that I must be close as the DC SPA seems to recieve traffic from the Roma unit..... :evil: |
| | |
| |||
| macudc I have made progress!!!!! successfully made and recieved calls!!! Check out this post. reply from sipura was helpfull. http://voxilla.com/index.php?name=PN...ht=point+point Let me know if that helps, cramer |
| |||
| cramer - did you succeed in this THROUGH natting firewalls? I truly am ready to send mine back because I canNOT get this to work. I followed to post, and was able to make calls on the LAN with no problems whatsoever. Then I moved the remote one (I am using a SPA2000) home behind a natting firewall (that has the correct ports open), and it no longer functions. I can call to the remote office, it rings, and when answered, there is no audio, and when hung up, the PSTN line fails to disconnect unless the line is physically pulled out of the SPA3000 at the office. I can not get a dial tone when I lift up the receiver connected to the SPA2000 in the remote office. The INFO page for the SPA3000(PSTN connection) shows 0 bytes/packets received for calls originating on the PSTN side. I'm dying here, and ready to give up, pending adequate documentation. I can't believe that the Sipura Users Guide doesn't cover this in detail! If anyone can help me, blessings be upon you, and many thanks! |
| |||
| Crammer, which method did you use the first from Montoya or the Sipura solution?? I am still having issues with method 2 but did not try it with staic public IPs. Am trying that now. Thanks for the link macudc |
| |||
| When I am trying to make the "bridged calls" the remote office (FXS) shows Mapped RTP Ports of 16xxx >> 16xxx. The PSTN line of the SPA3000 shows 16xxx >> 0. Reading through the posts, I figured it might have been a NAT problem, so I moved the FXO (SPA3000) outside the firewall, gave it a static outside IP on the T-1. Same result; 0 bytes received and the Mapped RTP Port for the FXS are 16xxx >> 16xxx and for the FXO 16xxx >> 0. I have basically the same config put forth by Sipura Technology in the above post. Please comment! THanks in advance |
| Thread Tools | |
| Display Modes | Rate This Thread |
| |
| | ||||
| Thread | Thread Starter | Forum | Replies | Last Post |
| Problem Solved (SPA3000 "fast busy" with <:@gw | DracoFelis | Linksys (Sipura) VoIP Support Forum | 7 | September 15th, 2006 12:25 PM |
| SPA3000: Voip to PSTN issue with "#" for PIN auth | jdeelizalde | Linksys (Sipura) VoIP Support Forum | 9 | July 31st, 2006 06:48 AM |
| [Help!] Spa-3K "Invalid Call State" all calls go t | jchuby | Linksys (Sipura) VoIP Support Forum | 1 | April 23rd, 2006 09:44 PM |
| make "free" cell calls via 3000/voip provider? | catrambull | Linksys (Sipura) VoIP Support Forum | 15 | September 8th, 2005 11:00 PM |
| SPA3000 constant "registration failed" errors | dgrove | VoicePulse Support Forum | 2 | September 12th, 2004 09:54 PM |