Failure to register SPA-3102 on "DSL-2750E" D-Link router

Similar Questions

• SPA-3102: unable to connect to the SIP server

  Hello

  We have a SPA-3102 installed and work properly in our US Office. We are trying to install another one in our office in India, but we cannot get this device to register with the SIP server.

  This device works perfectly when try us two different residences here in India with two different providers. But when we try to our office with a third party service provider, it is impossible to save.

  The internet connection at the office from the East through an ADSL router which is mode bridged with two IP static coming out of it. We are able to have access to the internet for our PC through this SPA-3102, but the SPA itself is unable to register with our SIP server. We use a Gizmo5 SIP server.

  The same parameters were working in our offices in the United States and two different residences in India but does not work in our office. I don't think that the ISP here has nothing to do with this problem. Would be - this works the ADSL router in bridge mode? Can I make changes in the settings of the SPA to work around this problem?

  Any help is appreciated.

  The problem could be the firewall or the router, but it is also possible that your Indian ISP is blocking voip connections.  It was reported that is not unknown in India.  Sometimes this type of blocking is done by throwing packages for voip standard sip signalling port 5060. You can easily change the SIP port in the SPA3102 to traffic returning to the SPA packages and in fact routers often changes the port without your knowledge, but packets destined for voip provider must use the port specified by the provider number.  Since you can access the internet with your pc, I would try to run tests to see if this is the case.

  I ran a few ping tests and tests to see if I could save with other providers or services that use replacing or other sip signaling ports.

• SPA-3102-backup/restore settings?

  Hello

  Is there a way to backup/restore the settings to a file in the SPA-3102 router?

  ASE

  There are several ways to save the settings of the SPA3102 according to some forums - check links ff

  http://forums.whirlpool.NET.au/Forum-replies-archive.cfm/756454.html

  http://Forum.Voxilla.com/Linksys-Sipura-VoIP-support-forum/SPA3102-compiler-configuration-backup-Mon...

• SPA 3102 Admin Guide

  Where can I download the administrator for the Linksys SPA 3102's Guide?

  I don't remember how many times I tried to find such a link and how many hours I have spent so far with it. I could find answers to this question, but the links are no longer valid.

  Thank you for your help.

  Try this link.

• DSL via a wireless router: slow XP, fast on Vista

  DSL via a wireless router: slow XP, fast on Vista

  Hello, I just moved into a new House that has DSL service. My Vista laptop connects seamlessly to the network, but my XP machine trickles in at 50-150 Kbps and is flaky. I rebooted everything, power all pedaled, verified and updated the drivers. (Maybe I missed something?)  The specs are all lower. Any thoughts on what is happening?

  Thank you!
  Brian

  EarthLink DSL
  VisionNet Modem
  Linksys wireless router (router broadband wireless - b)
  -BEFW11S4

  2 wireless laptops

  Get about 1 Mbps on a Vista Machine (which I believe is the expected rate).
  (Dell XPS M1210)
  Works with either ethernet or wireless connection to router

  Get about 50 Kbps on the XP Machine
  (Dell Latitude D830)
  Works with no wireless or ethernet connection to the router
  Intel Wireless Wifi Link 4965AGN,
  Driver: 12.4.3.9, Date: 28/05/2009

  Two laptop computers were working well 4 days ago in my previous House, sharing a connection to broadband Charter via a linksys wireless router.  The works of XP machine still work well pick up public Wi - Fi in coffee shops, etc.

  Under the name Vista computer works fine... I suggest to check again the Xp computer and disable the firewall and antivirus on the computer and try... You can try changing the wireless channel on the router... Also keep the mixed mode network mode.

• help with SPA 3102 (question graphcal)

  HI guys here is my situation (I draw so it would be easier for future reference):

  

  I want to pick up A phone and dial the Ext 101 101, 102 for Ext 102 and so on.
  also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.

  any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.

  castro69 wrote:
  also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.
  any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.

  I guess that's an analog PBX, otherwise you wouldn't need the SPA3102 through the internet.

  For communications between the SPA3102, I would use direct ip call, using the external ip address and the sip port numbers.  Think that the SPA3102 is two separate cards inside the box where treat you everyone with its sip port number added to the external ip address common.

  I would setup port sip distinctive numbers on each of the baths to keep things straight.  You have a number of separate port for tabs of line 1 and line PSTN.  You will need to send these to the SPA3102 adapter port numbers in their respective routers or firewall of the router will reject incoming packets on the internet.  I would also convey the port range rtp for voice, flow packs.
  On each tab of the line 1 and RTC, you define NAT Mapping Enable: YES, not record, make call without Reg Yes, years call without Reg No. I put your external ip address on the Sip tab under EXT IP.  This will tell the SPA3102 to use this address in the sip signaling. I assume you are using static external ip addresses.   On each tab of the line 1 you would activate IP Dial Yes.

  The analog PBX is connected to the FXO port on one of the Spa.  You should check the voltage level hung up and won and then set the line parameter usage on the RTC of the SPA line tab to halfway between the two readings.  You can read the levels of tension on the PSTN line tab.  Calls to the PBX of the PSTN line tab will go through the voip to PSTN gateway.  I set up the catwalk with http authentication and configure a user name and password.

  Details are starting to become quite complicated.  I'd get running through steps.  Get a job step before moving on to the next step.

  The 1st step would be to get A phone call/receive calls to a PBX.  You can configure the line 1 for FXS phone attached A to use port location PSTN 2 as the proxy using http authentication, and you can then dial the extensions you want to call.  Location 1 SPA3102 will send a guest of the sip Protocol to the tab location 2 SPA3102 from pstn line and the SPA3102 will dial the number on the FXO port to the PBX.

  For calls coming from the other direction of a PBX to slot 2 SPA3102 the only place where you can connect a voip call is in the SPA3102 numbering plan.  If you want to call only phone that is easy, install you just dialers-messengers automatic telephone in the pstn-to-voip dial plan.

  I'm not clear about what you want to do with phone B I take is Extension 104.

  I like your designs.  Can save a lot of words.

• SPA-3102: how to fill the calls from SIP to PSTN and vice versa?

  Hello, since my ISP to my office blocked SIP ports, of I'll try and use it at home where the ATA works very well.

  If I call you ATA installed in my office in the United States (where there is no problem) to my ATA at home in India, I have how to route the call selectively to theFXS port or port FXO of the anti-terrorism Act in India? I I want to answer the call directly using the phone connected to the ATA instrument or make another local call out on the PSTN connection.

  TIA

  You get two accounts of Gizmo.  You put an account on the tab line 1, you put the other account on the PSTN line tab.  If you want to ring the phone attached to the SPA3102, you dial the account tab line 1.  If you want to get the tone to fill an outgoing line PSTN call, you call the account registered on the PSTN line tab.

• SPA-3102 PSTN idle status, will not answer but watch sounds.

  Greetings,

  I am a complete newbie in the world of voip. I have an spa3102 and a computer virtual trixbox. When I use my softphone to compose, it makes to the spa3102 (shows the id of the last and gives a status of the appellant voip 'reply') but I get an "all circuits are busy now" and does not change the status of IDLE pstn. It also will not answer the phone for incoming calls to the PSTN.

  Outgoing stuff could be a problem of numbering plan (don't know not just guessing) here it is:

  (S0:xx. <@gw0>)

  I tried the regular (or default) (xx).

  incoming calls are supposed to go to a handset on my desk, for example ext 101 is the dial plan: (S0:101@internalip)

  I can't see the forest for the trees. Can someone give me some advice on where to look. I'm completely puzzled by something non-response externally, and why I get a all circuits are busy if there are inactive on the status page.

  an interesting note: If line1 is connected to an analog phone, it will dial a number, but always shows failover for a reason any...

  Thank you very much!!!

  Greetings!

  Thanks for the reply. I do not know if I explained properly (sounds like I don't have). I have a trixbox and I use the SPA in interface with my RTC and I want to have a fax machine (but later) on the FXS port. The problem I have is that I could not field calls to the collection. They would just ring and ring. I thought it was a dial plan (and hope it was not a hardware problem), but he just wouldn't answer the phone. He would show 'Ringing' like the PSTN status, but Voip would be "IDLE". I've been pouring through the guide of the administrator, forum messages for about 2 weeks now.

  But I didn't know what the problem was. The spa3102 is waiting until she can establish a connection (with the internal phone system) until it meets the PSTN. This is why it continues to ring. The problem why it shows it is inactive and (even with enabled full logging to a syslog box) no traffic to SIP the trixbox internal... because I had the port internal plugged in... not the WAN port (or internet). Thats right... silly me...

  I gave the wan port an address on the same subnet and moved the cable to it, once I could talk to her I change address internal to something even not on my network (if no routing problem) and left the network adapter disconnected (internal nic). Then I changed the wan to the old internal address address (trix trunk was already configured in this ip address anyway...) and PRESTO! I tried calling the number RTC and started to receive log entries showing the sip signaling and he's trying to route the call to the box of trix. It works now! So he was not answering the phone in my case, because it could not connect to the pbx or the ip phone and that because I don't use the WAN/INTERNET port on my SPA. Something I don't remember seeing in the textbooks as a "hafto' or a 'must '... I might have forgotten that of course, but I don't remember seeing him there.

  So if you enable logging, and then call this number (I used the PuTTY and 'tail-f /var/log/spa3102.log"to see the entries that they light up the box of syslog) and you don't see any sip traffic, it could be that you must use the WAN/INTERNET port instead of the internal port apparently logical... BE sure THAT ENABLE YOU REMOTE ADMIN if you decide to give it a shot.

  Also you can do the spa to answer the phone, even if it has no where to send the call (or if you want incoming people to use a PIN) on the GATEWAY PSTN - VOIP, there is a setting "off hook during the VOIP call" (default = no), but change that Yes - he would answer... and then sit there... If I had a PIN with a dial plan that is associated with this would give a fast busy after the entry of the PIN. BUT now everything is cool!

  Thank you!

• SPA 3102 - call "RTC for VOIP" setting is not on CLI/CID phone line active.

  Hi guys

  Recently, I discovered if the PSTN line is having with CLI/CID then helped the PSTN to VOIP Call not established with the standard settings. But others see (VOIP to PSTN) works well OK. Basically, it works well with the PSTN without CLI/CID lines. I guess when with CLI, there different voltage levels online. ?  You would let me know what are the parameters that I need to SPA in order to work with the PSTN line that allowed the CLI/CID please? This situation is covered by Sri Lanka Telecom PSTN lines. At the moment I have IT of standard parameters such as it comes with the SPA3102.

  Your feedback and help here...

  Syslog & debugging can save so much time...

  Line with no CLI:

   INVITE sip:[email protected]/* */ SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-a16a89a0 From: PradeephSL [email protected]/* */>;tag=4cc44dbb58f1a944o1 To: [email protected]/* */> Remote-Party-ID: PradeephSL [email protected]/* */>;screen=yes;party=calling Call-ID: [email protected]/* */ CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="17772695735" realm="callcentric.com" nonce="684082550c6f8ce50da482371a591df7" uri="sip:[email protected]/* */" algorithm=MD5 response="01c15ae601d9d9f5af10023f908a1a4c" opaque="" Contact: PradeephSL [email protected]/* */:5061> Expires: 240 User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 441 Allow: ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER Supported: x-sipura replaces Content-Type: application/sdp ... 

  This INVITATION is accepted by proxy.

  Line with CLI:

   INVITE sip:[email protected]/* */ SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-ea75ea0 From: PradeephSL [email protected]/* */>;tag=2d4fafa5d94faa2co1 To: [email protected]/* */> Remote-Party-ID: PradeephSL [email protected]/* */>;screen=yes;party=calling Call-ID: [email protected]/* */ CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="17772695735" realm="callcentric.com" nonce="fd5a5e637b6e56fea56886ee25aababa" uri="sip:[email protected]/* */" algorithm=MD5 response="b6b1aa7104d1be764de5c74f369ef5be" opaque="" Contact: PradeephSL [email protected]/* */:5061> Expires: 240 User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 441 Allow: ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER Supported: x-sipura replaces Content-Type: application/sdp ...

  This INVITATION is rejected by proxy with "403 incorrect authentication."

  But Howard has already struck and explained...

• ID (CLI of the incoming caller SPA-3102) truncates the last digit when the telephone number is longer.

  Hi team

  Depending on the subject, when the number of the caller is Longer (i.e. International call with Country code etc.) as 11 digits, it truncates the last digit of the incoming phone number. See picture attached. Full number is 60126140235, but he loses the last digit is 5. Any suggestions?

  Also, there are comma (,) before the number that consumes valuable space. How elimiante that?

   FXO:Start CNDD fxo cnddwrap_feed parse ok 0060126140 status=2 -- Caller ID: -- Name = (null) -- Remote Number = 0060126140 -- Dialable Number = (null) -- No Number Reason = (null) -- No Name Reason = (null) -- Message Waiting = (null) -- Date and Time = 07/22 18:06 FXO:CNDD name=, number=0060126140 FXO:Stop CNDD FXO:CNDD Name= Phone=0060126140

  According to the newspaper that you have provided, the number of callers number claimed 0060126140.

  Length maximum support of the number in the E.164 format is 12 digits, including the country code. SPA don't know any used number format, so it has no reason to truncate 10-digit - I guess that he supported at LEAST 12 E.164 numbers.

  As the number seems to be broken on the side operators (00 superfluous as prefix, truncated to 10 digits) there's nothing you can to with him on the side SPA. We cannot guess figures sent by the operator to you...

  I can explain what is happening even on the side of the operators (although I'm only guessing) - I guess that we are talking of two digit country code country, so 10 digits is the maximum length of the national number. It seems your operator to consider the number national number and truncate them to 10 numbers on their side.

• Failure to register product

  Hello

  Software: PC laptop Toshiba Regsitration
  Language: German
  Product: S11 - 11H

  I can't save my laptop during, enter the date to purhase. Whenever sending fails in recognizing good trained date format.

  Try saving your laptop directly on the Toshiba support page under http://eu.computers.toshiba-europe.com/innovation/generic/services_warranty_reg_map_eu/

  The registration status you can check the details page of unit - http://www.toshiba-tro.de/unitdetails

  Good bye

• TRUNK VOIP CREDITS SPA 3102

  
  

  reylon wrote:

• Failure to register files

  Whenever I'm editing a document and save as PDF, the program freezes and shows a blank screen. I get a dialog box saying "Adobe Acrobat DC has stopped working". After "Windows is looking for a solution... ». Which may be responsible for this, and how can I get on it? It costs money to not have this difficulty. Please notify.

  Hi dayob99,

  Please install the patch from this link https://helpx.adobe.com/acrobat/release-note/acrobat-dc-august-06-2015.html

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  Let me know if the problem persists.

  Kind regards

  Nicos

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  Hello

  Recently, I was doing the following error on our vCenter Server server while trying to import a virtual machine - "unable to connect to the server vCenter Converter Server.

  I followed the recommendations of the other post of the community http://communities.VMware.com/thread/239203 However, I still get the same error.

  I tried to run a repair on vCenter Converter of add/remove programs however I get the following error:

  "Cannot save the extension.

  I restarted the server as well and still no change.

  Any ideas?

  If you have changed the IP address on the VC, your best option is probably uninstall the Update Manager and converter, once again.  delete their directories under \program files\vmware\infrastructure\ installation, restart and install again.

• Problem with access to the camera on the net via dsl modem and linksys router

  OK, I read all posts here on what to do in this case and think I have tried everything. I have an Actiontec dsl (verizon) and a v2 wrt110 router. I can see the camera locally, but not on the net. I had it hooked to another place but with cable service broadband with no problems.

  I think that the problem of my lie with ip addresses.

  Camera 192.168.0.115

  Router 192.168.0.1

  Modem 192.168.1.1

  The port forwards on the modem MUST be set to port forward to the IP Address of the Linksys router. It should be only the modem IP can see, so it's the only place you can move forward from the modem. What exactly is in this drop-down list in the config of the modem port forwarding?

  The problem is that there are two distinct here networks once you are inside your modem. There is a network between the modem and the router. Allthat is on it is the modem's ethernet interface and the port of the Internet router (or WAN). There is another network on the side the router as your PC's LAN, camera, etc. are on. All know the modem is how to get to the network to which it is directly connected to. While you could add a route staic on the modem to tell him how to get to this network, the modem must still know I have oprt route traffic to your camera to the router.